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Changelog for asterisk-voicemail-odbc-13.7.1-1.fc22.i686.rpm :
Thu Feb 4 13:00:00 2016 Jared Smith - 13.7.1-1
- Update to upstream 13.7.1 release for security fixes
- Resolves AST-2016-001: BEAST vulnerability in HTTP server
- Resolves AST-2016-002: File descriptor exhaustion in chan_sip
- Resolves AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data
- Full changelog at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
- Also build the \'radius\' sub-package against freeradius-client-devel, as the
radiusclient-ng project is dead
Wed Feb 3 13:00:00 2016 Fedora Release Engineering - 13.3.2-3.1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild
Mon Jan 25 13:00:00 2016 Jared Smith - 13.3.2-3
- Remove %defattr macro invocations, as they are no longer needed
Sat Jan 23 13:00:00 2016 Robert Scheck - 13.3.2-2
- Rebuild for libical 2.0.0
Wed Jun 17 14:00:00 2015 Fedora Release Engineering - 13.3.2-1.2
- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild
Sat Jun 6 14:00:00 2015 Jitka Plesnikova - 13.3.2-1.1
- Perl 5.22 rebuild
Thu Apr 9 14:00:00 2015 Jeffrey C. Ollie - 13.3.2-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
-
* AST-2015-003: TLS Certificate Common name NULL byte exploit
-
- When Asterisk registers to a SIP TLS device and and verifies the server,
- Asterisk will accept signed certificates that match a common name other than
- the one Asterisk is expecting if the signed certificate has a common name
- containing a null byte after the portion of the common name that Asterisk
- expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
Thu Apr 9 14:00:00 2015 Jeffrey C. Ollie - 13.3.1-1:
- The Asterisk Development Team has announced the release of Asterisk 13.3.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 13.3.1 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
-
* --- pjsip: resolve compatibility problem with ast_sip_session
- (Closes issue ASTERISK-24941. Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1
Wed Apr 1 14:00:00 2015 Jeffrey C. Ollie - 13.3.0-1:
- The Asterisk Development Team has announced the release of Asterisk 13.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 13.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
* ASTERISK-24703 - ARI: Add the ability to \"transfer\" (redirect) a
- channel (Reported by Matt Jordan)
-
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
- (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
- string copy (Reported by Yura Kocyuba)
-
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
- sorcery.conf false ERROR messages may occur (Reported by Joshua
- Colp)
-
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
- (Reported by Matt Jordan)
-
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
- res_odbc (Reported by ibercom)
-
* ASTERISK-24479 - Enable REF_DEBUG for module references
- (Reported by Corey Farrell)
-
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
- fully disconnect underlying socket, leading to events being
- dropped with no additional information (Reported by Matt Jordan)
-
* ASTERISK-24772 - ODBC error in realtime sippeers when device
- unregisters under MariaDB (Reported by Richard Miller)
-
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
- is destroyed by ARI during shutdown (Reported by Richard
- Mudgett)
-
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
- by Zane Conkle)
-
* ASTERISK-24015 - app_transfer fails with PJSIP channels
- (Reported by Private Name)
-
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
- transfer scenario. (Reported by Mark Michelson)
-
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
- Niklas Larsson)
-
* ASTERISK-24716 - Improve pjsip log messages for presence
- subscription failure (Reported by Rusty Newton)
-
* ASTERISK-24612 - res_pjsip: No information if a required sorcery
- wizard is not loaded (Reported by Joshua Colp)
-
* ASTERISK-24768 - res_timing_pthread: file descriptor leak
- (Reported by Matthias Urlichs)
-
* ASTERISK-24685 - \"pjsip show version\" CLI command (Reported by
- Joshua Colp)
-
* ASTERISK-24632 - install_prereq script installs pjproject
- without IPv6 support (Reported by Rusty Newton)
-
* ASTERISK-24085 - Documentation - We should remove or further
- document the \'contact\' section in pjsip.conf (Reported by Rusty
- Newton)
-
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
- JoshE)
-
* ASTERISK-24700 - CRASH: NULL channel is being passed to
- ast_bridge_transfer_attended() (Reported by Zane Conkle)
-
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
- (Reported by Corey Farrell)
-
* ASTERISK-24799 - [patch] make fails with undefined reference to
- SSLv3_client_method (Reported by Alexander Traud)
-
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
- Events (Reported by klaus3000)
-
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
- call (Reported by Marcel Manz)
-
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
- (Reported by Panos Gkikakis)
-
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
- for playing back messages stored in IMAP - play_message: No
- origtime (Reported by Graham Barnett)
-
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
- OSX with 64 bit integers (Reported by Corey Farrell)
-
* ASTERISK-24796 - Codecs and bucket schema\'s prevent module
- unload (Reported by Corey Farrell)
-
* ASTERISK-24724 - \'httpstatus\' Web Page Produces Incomplete HTML
- (Reported by Ashley Sanders)
-
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
- is invalid (Reported by Rusty Newton)
-
* ASTERISK-24785 - \'Expires\' header missing from 200 OK on
- REGISTER (Reported by Ross Beer)
-
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful
- response on non-existent variable (Reported by Joshua Colp)
-
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
- (Reported by Kevin Harwell)
-
* ASTERISK-24812 - ARI: Creating channels through /channels
- resource always uses SLIN, which results in unneeded transcoding
- (Reported by Matt Jordan)
-
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
- thread ID being passed to pthread_kill (Reported by JoshE)
-
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
- fail (Reported by Terry Wilson)
-
* ASTERISK-23214 - chan_sip WARNING message \'We are requesting
- SRTP for audio, but they responded without it\' is ambiguous and
- wrong in some cases (Reported by Rusty Newton)
-
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
- error response and BYE are sent to the caller (Reported by
- Makoto Dei)
-
* ASTERISK-18105 - most of asterisk modules are unbuildable in
- cygwin environment (Reported by feyfre)
-
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-
* ASTERISK-24751 - Integer values in json payload to ARI cause
- asterisk to crash (Reported by jeffrey putnam)
-
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
- building a peer causes a peer poke during request handling
- (Reported by Richard Mudgett)
-
* ASTERISK-24825 - Caller ID not recognized using
- Centrex/Distinctive dialing (Reported by Richard Mudgett)
-
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
- HAVE_PJPROJECT (Reported by Stefan Engström)
-
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
- (Reported by Kevin Harwell)
-
* ASTERISK-24755 - Asterisk sends unexpected early BYE to
- transferrer during attended transfer when using a Stasis bridge
- (Reported by John Bigelow)
-
* ASTERISK-24739 - [patch] - Out of files -- call fails --
- numerous files with inodes from under /usr/share/zoneinfo,
- mostly posixrules (Reported by Ed Hynan)
-
* ASTERISK-23390 - NewExten Event with application AGI shows up
- before and after AGI runs (Reported by Benjamin Keith Ford)
-
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
- voicemail stored in LDAP (Reported by Graham Barnett)
-
* ASTERISK-24808 - res_config_odbc: Improper escaping of
- backslashes occurs with MySQL (Reported by Javier Acosta)
-
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
- by Anatoli)
-
* ASTERISK-20850 - [patch]Nested functions aren\'t portable.
- Adapting RAII_VAR to use clang/llvm blocks to get the
- same/similar functionality. (Reported by Diederik de Groot)
-
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
- connection on error (Reported by Dmitriy Serov)
-
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
- by Frank DiGennaro)
-
* ASTERISK-21038 - Bad command completion of \"core set debug
- channel\" (Reported by Richard Kenner)
-
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
- Dave Cabot)
-
* ASTERISK-16779 - Cannot disallow unknown format \'\' (Reported by
- Atis Lezdins)
-
* ASTERISK-24876 - Investigate reference leaks from
- tests/channels/local/local_optimize_away (Reported by Corey
- Farrell)
-
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
- by Corey Farrell)
-
* ASTERISK-24817 - init_logger_chain: unreachable code block
- (Reported by Corey Farrell)
-
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
- snuffy)
-
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
- under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
- (Reported by Ben Merrills)
-
* ASTERISK-24811 - asterisk-publication sorcery object does not
- use realtime (Reported by Matt Hoskins)
-
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
- Couldn\'t find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0
Wed Apr 1 14:00:00 2015 Jeffrey C. Ollie - 13.2.0-1:
- The Asterisk Development Team has announced the release of Asterisk 13.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 13.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
- all at the same time. (Reported by Richard Mudgett)
-
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
- when using non-default sorcery wizard (Reported by Kevin
- Harwell)
-
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
- from JSSIP (Reported by Badalian Vyacheslav)
-
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
- media streams results in 488 (Reported by Matt Jordan)
-
* ASTERISK-24563 - Direct Media calls within private network
- sometimes get one way audio (Reported by Kevin Harwell)
-
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
- race condition in accessing codec in stored ast_frame and codec
- core (Reported by Matt Jordan)
-
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
- enabled (Reported by Richard Mudgett)
-
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
- enabled (Reported by Andreas Steinmetz)
-
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
- casts char to unsigned int (Reported by Walter Doekes)
-
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
- channel (Reported by Niklas Larsson)
-
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
- chosen for RTP compatible channels when the DTMF mode is not
- compatible (Reported by Yaniv Simhi)
-
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
- level - \'Remote address is null, most likely RTP has been
- stopped\' (Reported by Rusty Newton)
-
* ASTERISK-24513 - Local channel apparently leaked in off-nominal
- DTMF attended transfer (Reported by Mark Michelson)
-
* ASTERISK-23733 - \'reload acl\' fails if acl.conf is not present
- on startup (Reported by Richard Kenner)
-
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
- destination when \'sendrpid=yes\' (in proxy environment) (Reported
- by Karsten Wemheuer)
-
* ASTERISK-23841 - DTMF atxfer doesn\'t set CallerID for the recall
- calls to the transferrer. (Reported by Richard Mudgett)
-
* ASTERISK-24376 - res_pjsip_refer: REFER request for remote
- session attempts to direct channel to external_replaces
- extension instead of context, without providing for the
- Referred-To SIP URI (Reported by Matt Jordan)
-
* ASTERISK-24591 - Stasis() side of an ARI originated channel
- cannot be Redirected (Reported by Kinsey Moore)
-
* ASTERISK-24049 - Asterisk Manager Interface: A number of list
- type responses aren\'t using astman_send_listack (Reported by
- Jonathan Rose)
-
* ASTERISK-24637 - Channel re-enters Stasis() when it should not
- (Reported by John Bigelow)
-
* ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
- not function (Reported by John Kiniston)
-
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
- (Reported by Kristian Høgh)
-
* ASTERISK-20744 - [patch] Security event logging does not work
- over syslog (Reported by Michael Keuter)
-
* ASTERISK-24665 - Configure check required for
- pjsip_get_dest_info() (Reported by Mark Michelson)
-
* ASTERISK-23850 - Park Application does not respect Return
- Context Priority (Reported by Andrew Nagy)
-
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
- in the CFlags returned (Reported by Diederik de Groot)
-
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
- while attempting to publish (Reported by Kevin Harwell)
-
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
- (Reported by Corey Farrell)
-
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
- on cross compilation (Reported by abelbeck)
-
* ASTERISK-24624 - Transfer to invalid extension results in hung
- channel. (Reported by Zane Conkle)
-
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
- Incorrect External Addresses is Used in SIP Packets When
- Responding to INVITE (Reported by David Justl)
-
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
- voicemail is not deleted after review, hangup (Reported by LEI
- FU)
-
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
- 32-bit packages on 64-bit hosts (Reported by Ben Klang)
-
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
- to most traffic, potential deadlock (Reported by Jeff Collell)
-
* ASTERISK-24560 - Creating a named ARI bridge twice causes a
- crash (Reported by Kinsey Moore)
-
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
- MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
- by Matt Jordan)
-
* ASTERISK-24640 - Registration pending stays forever after sip
- reload (Reported by Max Man)
-
* ASTERISK-24673 - outgoing sip registers cannot be removed or
- modified without doing restart (or doing module unload
- chan_sip.so) (Reported by Stefan Engström)
-
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
- m() option does not queue an MWI event (Reported by Gareth
- Palmer)
-
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
- fails to get app name (Reported by John Bigelow)
-
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
- column comparison for \'defaultuser\' (Reported by
- HZMI8gkCvPpom0tM)
-
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
- (Reported by Kevin Harwell)
-
* ASTERISK-24626 - Voicemail passwords not being stored in ARA
- (Reported by Paddy Grice)
-
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
- in bridge_channel.c (Reported by George Joseph)
-
* ASTERISK-24544 - Compile fails on OSX Yosemite because of
- incorrect detection of htonll and ntohll (Reported by George
- Joseph)
-
* ASTERISK-24723 - confbridge: CLI command \'confbridge list XXXX\'
- no longer displays user menus (Reported by Matt Jordan)
-
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
- \'module not found\' during a Reload operation (Reported by Matt
- Jordan)
-
* ASTERISK-24719 - ConfBridge recording channels get stuck when
- recording started/stopped more than once (Reported by Richard
- Mudgett)
-
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
- by Kevin Harwell)
-
* ASTERISK-24728 - tcptls: Bad file descriptor error when
- reloading chan_sip (Reported by Kevin Harwell)
-
* ASTERISK-24729 - Outbound registration not occuring on new
- registrations after reload. (Reported by Richard Mudgett)
-
* ASTERISK-24676 - Security Vulnerability: URL request injection
- in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
-
* ASTERISK-24666 - Security Vulnerability: RTP not closed after
- sip call using unsupported codec (Reported by Y Ateya)
-
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
- versions (Reported by Jared Biel)
-
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
- Stephan Eisvogel)
-
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
-
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
- is ever received (Reported by Marco Paland)
-
* ASTERISK-24737 - When agent not logged in, agent status shows
- unavailable, queue status shows agent invalid (Reported by
- Richard Mudgett)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-24552 - ARI: Allow associating a channel as an
- initiator of an Origination for record keeping purposes
- (Reported by Matt Jordan)
-
* ASTERISK-24553 - ARI/AMI: Include language in standard channel
- snapshot output (Reported by Matt Jordan)
-
* ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
- Matt Jordan)
-
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
- connection-oriented transports. (Reported by Matt Jordan)
-
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
- handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
- Israel))
-
* ASTERISK-24678 - [PATCH] Added atxfer
* settings to
- features.conf.sample (Reported by Niklas Larsson)
-
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
- by cloos)
-
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
- Dan Jenkins)
-
* ASTERISK-24316 - For httpd server, need option to define server
- name for security purposes (Reported by Andrew Nagy)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0
Fri Jan 30 13:00:00 2015 Jeffrey C. Ollie - 13.1.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
-
* AST-2015-001: File descriptor leak when incompatible codecs are offered
-
- Asterisk may be configured to only allow specific audio or
- video codecs to be used when communicating with a
- particular endpoint. When an endpoint sends an SDP offer
- that only lists codecs not allowed by Asterisk, the offer
- is rejected. However, in this case, RTP ports that are
- allocated in the process are not reclaimed.
-
- This issue only affects the PJSIP channel driver in
- Asterisk. Users of the chan_sip channel driver are not
- affected.
-
-
* AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
- CVE-2014-8150 reported an HTTP request injection
- vulnerability in libcURL. Asterisk uses libcURL in its
- func_curl.so module (the CURL() dialplan function), as well
- as its res_config_curl.so (cURL realtime backend) modules.
-
- Since Asterisk may be configured to allow for user-supplied
- URLs to be passed to libcURL, it is possible that an
- attacker could use Asterisk as an attack vector to inject
- unauthorized HTTP requests if the version of libcURL
- installed on the Asterisk server is affected by
- CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
Fri Jan 30 13:00:00 2015 Jeffrey C. Ollie - 13.1.0-1
- The Asterisk Development Team has announced the release of Asterisk 13.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 13.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
* ASTERISK-24554 - AMI/ARI: Generate events on connected line
- changes (Reported by Matt Jordan)
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
- against libsrtp-1.5.0 (Reported by Patrick Laimbock)
-
* ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
- Corey Farrell)
-
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
- leak (Reported by Corey Farrell)
-
* ASTERISK-24430 - missing letter \"p\" in word response in
- OriginateResponse event documentation (Reported by Dafi Ni)
-
* ASTERISK-24437 - Review implementation of ast_bridge_impart for
- leaks and document proper usage (Reported by Scott Griepentrog)
-
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
- Corey Farrell)
-
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
- Corey Farrell)
-
* ASTERISK-24458 - chan_phone fails to build on big endian systems
- (Reported by Tzafrir Cohen)
-
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
-
* ASTERISK-24304 - asterisk crashing randomly because of unistim
- channel (Reported by dhanapathy sathya)
-
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
- Nick Adams)
-
* ASTERISK-24462 - res_pjsip: Stale qualify statistics after
- disablementation (Reported by Kevin Harwell)
-
* ASTERISK-24465 - audiohooks list leaks reference to formats
- (Reported by Corey Farrell)
-
* ASTERISK-24466 - app_queue: fix a couple leaks to struct
- call_queue (Reported by Corey Farrell)
-
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
- (Reported by Corey Farrell)
-
* ASTERISK-24411 - [patch] Status of outbound registration is not
- changed upon unregistering. (Reported by John Bigelow)
-
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
- leaks (Reported by Corey Farrell)
-
* ASTERISK-24480 - res_http_websockets: Module reference decrease
- below zero (Reported by Corey Farrell)
-
* ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
- audiohook callback (Reported by Corey Farrell)
-
* ASTERISK-24487 - configuration: sections should be loadable as
- template even when not marked (Reported by Scott Griepentrog)
-
* ASTERISK-20127 - [Regression] Config.c config_text_file_load()
- unescapes semicolons (\"\\;\" -> \";\") turning them into comments
- (corruption) on rewrite of a config file (Reported by George
- Joseph)
-
* ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
- when DNS settings invalid (Reported by Melissa Shepherd)
-
* ASTERISK-24307 - Unintentional memory retention in stringfields
- (Reported by Etienne Lessard)
-
* ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
- Conkle)
-
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
- extra calls to ast_module_unref (Reported by Corey Farrell)
-
* ASTERISK-24447 - Bridge DTMF hooks: Audio doesn\'t pass when
- waiting for more matching digits. (Reported by Richard Mudgett)
-
* ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
- queue caller (Reported by Steve Pitts)
-
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
- (Reported by Corey Farrell)
-
* ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
- header fix (Reported by abelbeck)
-
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
- length exceeds 50 (roughly) national symbols (Reported by
- Dmitriy Bubnov)
-
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
- revision r227276 (Reported by Xavier Hienne)
-
* ASTERISK-24505 - manager: http connections leak references
- (Reported by Corey Farrell)
-
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
- coverage are enabled (Reported by Corey Farrell)
-
* ASTERISK-24444 - PBX: Crash when generating extension for
- pattern matching hint (Reported by Leandro Dardini)
-
* ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
- packet to JSON for res_hep_rtcp and report blocks are greater
- than 1 (Reported by Gregory Malsack)
-
* ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
- transfer (Reported by Beppo Mazzucato)
-
* ASTERISK-24501 - ARI: Moving a channel between bridges followed
- by a hangup can cause an ARI client to not receive an expected
- ChannelLeftBridge event before StasisEnd (Reported by Matt
- Jordan)
-
* ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
- (Reported by Leon Rowland)
-
* ASTERISK-23651 - Reloading some modules that are loaded already,
- results in \'No such module\' before a successful reload (Reported
- by Rusty Newton)
-
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
- endmarked users when last marked user leaves (Reported by Matt
- Jordan)
-
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
- (Reported by David Woolley)
-
* ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
- with \"400 bad request\" - DEBUG shows \"Received a REFER without a
- parseable Refer-To\" (Reported by Beppo Mazzucato)
-
* ASTERISK-24535 - stringfields: Fix regression from fix for
- unintentional memory retention and another issue exposed by the
- fix (Reported by Corey Farrell)
-
* ASTERISK-24471 - Crash - assert_fail in libc in
- pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
- (Reported by yaron nahum)
-
* ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
- in-dialog with invalid target causes crash (Reported by Joshua
- Colp)
-
* ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
- module load (Reported by Matt Jordan)
-
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
- allow blocked addresses through (Reported by Matt Jordan)
-
* ASTERISK-24542 - [patch]Failure showing codecs via \'core show
- channeltype \' (Reported by snuffy)
-
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
- by xrobau)
-
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
- voicemail under high concurrency with an IMAP backend (Reported
- by David Duncan Ross Palmer)
-
* ASTERISK-24572 - [patch]App_meetme is loaded without its
- defaults when the configuration file is missing (Reported by
- Nuno Borges)
-
* ASTERISK-24573 - [patch]Out of sync conversation recording when
- divided in multiple recordings (Reported by Nuno Borges)
-
* ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
- reliably transmitted during transfers (Reported by Matt Jordan)
-
* ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
- extension to another pjsip extension (Reported by Abhay Gupta)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
- property \'unanswered\' (Reported by Matt Jordan)
-
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
- column in the cel_odbc module (Reported by Etienne Lessard)
-
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
- recordings (Reported by Ben Smithurst)
-
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
- lookups (Reported by Birger \"WIMPy\" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0
Thu Jan 29 13:00:00 2015 Peter Robinson 13.0.2-3
- Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021
Thu Jan 15 13:00:00 2015 Tom Callaway - 13.0.2-2
- update for lua 5.3
Wed Dec 10 13:00:00 2014 Jeffrey C. Ollie - 13.0.2-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
-
* AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
- When handling a WebSocket frame the res_http_websocket module dynamically
- changes the size of the memory used to allow the provided payload to fit. If a
- payload length of zero was received the code would incorrectly attempt to
- resize to zero. This operation would succeed and end up freeing the memory but
- be treated as a failure. When the session was subsequently torn down this
- memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
Thu Nov 20 13:00:00 2014 Jeffrey C. Ollie - 13.0.1-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
- 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
-
* AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
- address families
-
- Many modules in Asterisk that service incoming IP traffic have ACL options
- (\"permit\" and \"deny\") that can be used to whitelist or blacklist address
- ranges. A bug has been discovered where the address family of incoming
- packets is only compared to the IP address family of the first entry in the
- list of access control rules. If the source IP address for an incoming
- packet is not of the same address as the first ACL entry, that packet
- bypasses all ACL rules.
-
-
* AST-2014-018: Permission Escalation through DB dialplan function
-
- The DB dialplan function when executed from an external protocol, such as AMI,
- could result in a privilege escalation. Users with a lower class authorization
- in AMI can access the internal Asterisk database without the required SYSTEM
- class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
-
* AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
- The ConfBridge application uses an internal bridging API to implement
- conference bridges. This internal API uses a state model for channels within
- the conference bridge and transitions between states as different things
- occur. Unload load it is possible for some state transitions to be delayed
- causing the channel to transition from being hung up to waiting for media. As
- the channel has been hung up remotely no further media will arrive and the
- channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
-
* AST-2014-017: Permission Escalation via ConfBridge dialplan function and
- AMI ConfbridgeStartRecord Action
-
- The CONFBRIDGE dialplan function when executed from an external protocol (such
- as AMI) can result in a privilege escalation as certain options within that
- function can affect the underlying system. Additionally, the AMI
- ConfbridgeStartRecord action has options that would allow modification of the
- underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
-
* AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
- The Asterisk module res_pjsip provides the ability to configure ACLs that may
- be used to reject SIP requests from various hosts. However, the module
- currently fails to create and apply the ACLs defined in its configuration
- file on initial module load.
-
-
* AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
- The chan_pjsip channel driver uses a queue approach for relating to SIP
- sessions. There exists a race condition where actions may be queued to answer
- a session or send ringing after a SIP session has been terminated using a
- CANCEL request. The code will incorrectly assume that the SIP session is still
- active and attempt to send the SIP response. The PJSIP library does not
- expect the SIP session to be in the disconnected state when sending the
- response and asserts.
-
-
* AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
- When handling an INVITE with Replaces message the res_pjsip_refer module
- incorrectly assumes that it will be operating on a channel that has just been
- created. If the INVITE with Replaces message is sent in-dialog after a session
- has been established this assumption will be incorrect. The res_pjsip_refer
- module will then hang up a channel that is actually owned by another thread.
- When this other thread attempts to use the just hung up channel it will end up
- using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
Thu Nov 20 13:00:00 2014 Jeffrey C. Ollie - 13.0.0-1
- The Asterisk Development Team is pleased to announce the release of
- Asterisk 13.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- Asterisk 13 is the next major release series of Asterisk. It is a Long Term
- Support (LTS) release, similar to Asterisk 11. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- For important information regarding upgrading to Asterisk 13, please see the
- Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13
-
- A short list of new features includes:
-
-
* Asterisk security events are now provided via AMI, allowing end users to
- monitor their Asterisk system in real time for security related issues.
-
-
* Both AMI and ARI now allow external systems to control the state of a mailbox.
- Using AMI actions or ARI resources, external systems can programmatically
- trigger Message Waiting Indicators (MWI) on subscribed phones. This is of
- particular use to those who want to build their own VoiceMail application
- using ARI.
-
-
* ARI now supports the reception/transmission of out of call text messages using
- any supported channel driver/protocol stack through ARI. Users receive out of
- call text messages as JSON events over the ARI websocket connection, and can
- send out of call text messages using HTTP requests.
-
-
* The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act
- as a Resource List Server. This includes defining lists of presence state,
- mailbox state, or lists of presence state/mailbox state; managing
- subscriptions to lists; and batched delivery of NOTIFY requests to
- subscribers.
-
-
* The PJSIP stack can now be used as a means of distributing device state or
- mailbox state via PUBLISH requests to other Asterisk instances. This is
- analogous to Asterisk\'s clustering support using XMPP or Corosync; unlike
- existing clustering mechanisms, using the PJSIP stack to perform the
- distribution of state does not rely on another daemon or server to perform the
- work.
-
- And much more!
-
- More information about the new features can be found on the Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation
-
- A full list of all new features can also be found in the CHANGES file:
-
- http://svnview.digium.com/svn/asterisk/branches/13/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0
Fri Nov 14 13:00:00 2014 Tom Callaway - 11.13.1-2
- rebuild for new libsrtp
Mon Oct 20 14:00:00 2014 Jeffrey C. Ollie - 11.13.1-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1,
- 11.13.1, 12.6.1, and 13.0.0-beta3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
-
* AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability
-
- Asterisk is susceptible to the POODLE vulnerability in two ways:
- 1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their
- encrypted connections.
- 2) The core TLS handling in Asterisk, which is used by the chan_sip channel
- driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by
- default allow a TLS connection to fallback to SSLv3. This allows for a
- MITM to potentially force a connection to fallback to SSLv3, exposing it
- to the POODLE vulnerability.
-
- These issues have been resolved in the versions released in conjunction with
- this security advisory.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-011, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert7
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.31.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.6.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta3
-
- The security advisory is available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-011.pdf
Mon Oct 20 14:00:00 2014 Jeffrey C. Ollie - 11.13.0-1
- The Asterisk Development Team has announced the release of Asterisk 11.13.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-24032 - Gentoo compilation emits warning:
- \"_FORTIFY_SOURCE\" redefined (Reported by Kilburn)
-
* ASTERISK-24225 - Dial option z is broken (Reported by
- dimitripietro)
-
* ASTERISK-24178 - [patch]fromdomainport used even if not set
- (Reported by Elazar Broad)
-
* ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
- warnings and ref leaks (Reported by Walter Doekes)
-
* ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
- ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
-
* ASTERISK-24019 - When a Music On Hold stream starts it restarts
- at beginning of file. (Reported by Jason Richards)
-
* ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
- if ever not able to resolve (Reported by David Herselman)
-
* ASTERISK-24211 - testsuite: Fix the dial_LS_options test
- (Reported by Matt Jordan)
-
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
- Mohod)
-
* ASTERISK-23577 - res_rtp_asterisk: Crash in
- ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
- Jay Jideliov)
-
* ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
- concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
- by Roman Skvirsky)
-
* ASTERISK-24301 - Security: Out of call MESSAGE requests
- processed via Message channel driver can crash Asterisk
- (Reported by Matt Jordan)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-24171 - [patch] Provide a manpage for the aelparse
- utility (Reported by Jeremy Lainé)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0
Mon Oct 20 14:00:00 2014 Jeffrey C. Ollie - 11.12.1-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11 and 12. The available security releases are
- released as versions 11.6-cert6, 11.12.1, and 12.5.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- Please note that the release of these versions resolves the following security
- vulnerability:
-
-
* AST-2014-010: Remote Crash when Handling Out of Call Message in Certain
- Dialplan Configurations
-
- Additionally, the release of Asterisk 12.5.1 resolves the following security
- vulnerability:
-
-
* AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests
-
- Note that the crash described in AST-2014-010 can be worked around through
- dialplan configuration. Given the likelihood of the issue, an advisory was
- deemed to be warranted.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-009 and AST-2014-010, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert6
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-009.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-010.pdf
Mon Oct 20 14:00:00 2014 Jeffrey C. Ollie - 11.12.0-1
- The Asterisk Development Team has announced the release of Asterisk 11.12.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
- empty string is a bit over zealous (Reported by Matt Jordan)
-
* ASTERISK-23985 - PresenceState Action response does not contain
- ActionID; duplicates Message Header (Reported by Matt Jordan)
-
* ASTERISK-23814 - No call started after peer dialed (Reported by
- Igor Goncharovsky)
-
* ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
- should not call sip_destroy (Reported by Corey Farrell)
-
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
- loaded, but dialplan not available (Reported by Dennis Guse)
-
* ASTERISK-18345 - [patch] sips connection dropped by asterisk
- with a large INVITE (Reported by Stephane Chazelas)
-
* ASTERISK-23508 - Memory Corruption in
- __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-21178 - Improve documentation for manager command
- Getvar, Setvar (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0
Mon Oct 20 14:00:00 2014 Jeffrey C. Ollie - 11.11.0-1
- The Asterisk Development Team has announced the release of Asterisk 11.11.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting
- at Invite, UAC starts counting at 200 OK. (Reported by i2045)
-
* ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
- by Peter Whisker)
-
* ASTERISK-23582 - [patch]Inconsistent column length in
*odbc
- (Reported by Walter Doekes)
-
* ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
- categories but the requested one (Reported by zvision)
-
* ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
- results in several bridges with same conf_name (Reported by
- Iñaki Cívico)
-
* ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
- AMI when waiting to enter a conference (Reported by Matt Jordan)
-
* ASTERISK-23683 - #includes - wildcard character in a path more
- than one directory deep - results in no config parsing on module
- reload (Reported by tootai)
-
* ASTERISK-23827 - autoservice thread doesn\'t exit at shutdown
- (Reported by Corey Farrell)
-
* ASTERISK-23609 - Security: AMI action MixMonitor allows
- arbitrary programs to be run (Reported by Corey Farrell)
-
* ASTERISK-23673 - Security: DOS by consuming the number of
- allowed HTTP connections. (Reported by Richard Mudgett)
-
* ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
- a DEBUG level of zero (Reported by Rusty Newton)
-
* ASTERISK-23766 - [patch] Specify timeout for database write in
- SQLite (Reported by Igor Goncharovsky)
-
* ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
- with Lua 5.2 or greater due to addition of goto statement
- (Reported by Rusty Newton)
-
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is
- loaded, but dialplan not available (Reported by Dennis Guse)
-
* ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
- length if ICE (Reported by Richard Kenner)
-
* ASTERISK-23790 - [patch] - SIP From headers longer than 256
- characters result in dropped call and \'No closing bracket\'
- warnings. (Reported by uniken1)
-
* ASTERISK-23917 - res_http_websocket: Delay in client processing
- large streams of data causes disconnect and stuck socket
- (Reported by Matt Jordan)
-
* ASTERISK-23908 - [patch]When using FEC error correction,
- asterisk tries considers negative sequence numbers as missing
- (Reported by Torrey Searle)
-
* ASTERISK-23921 - refcounter.py uses excessive ram for large refs
- files (Reported by Corey Farrell)
-
* ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
- objects that were already freed (Reported by Corey Farrell)
-
* ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
- between attributes (Reported by Alexander Traud)
-
* ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
- (Reported by Steve Davies)
-
* ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
- PI) in revision 413765 breaks working environments (Reported by
- Pavel Troller)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-23492 - Add option to safe_asterisk to disable
- backgrounding (Reported by Walter Doekes)
-
* ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
- (Reported by Jay Jideliov)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0
Thu Aug 28 14:00:00 2014 Jitka Plesnikova - 11.10.2-2.2
- Perl 5.20 rebuild
Fri Aug 15 14:00:00 2014 Fedora Release Engineering - 11.10.2-2.1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild
Thu Jun 19 14:00:00 2014 Jeffrey Ollie - 11.10.2-2:
- Drop the 389 directory server schema (1061414)
Thu Jun 19 14:00:00 2014 Jeffrey Ollie - 11.10.2-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
- releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2,
- and 12.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- These releases resolve security vulnerabilities that were previously fixed in
- 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1. Unfortunately, the fix
- for AST-2014-007 inadvertently introduced a regression in Asterisk\'s TCP and TLS
- handling that prevented Asterisk from sending data over these transports. This
- regression and the security vulnerabilities have been fixed in the versions
- specified in this release announcement.
-
- The security patches for AST-2014-007 have been updated with the fix for the
- regression, and are available at http://downloads.asterisk.org/pub/security
-
- Please note that the release of these versions resolves the following security
- vulnerabilities:
-
-
* AST-2014-005: Remote Crash in PJSIP Channel Driver\'s Publish/Subscribe
- Framework
-
-
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
- Shell Access
-
-
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
- Connections
-
-
* AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008,
- which were released with the previous versions that addressed these
- vulnerabilities.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert7
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.2
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-005.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-008.pdf
Thu Jun 19 14:00:00 2014 Jeffrey Ollie - 11.10.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
- releases are released as versions 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1,
- and 12.3.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following issue:
-
-
* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
- Connections
-
- Establishing a TCP or TLS connection to the configured HTTP or HTTPS port
- respectively in http.conf and then not sending or completing a HTTP request
- will tie up a HTTP session. By doing this repeatedly until the maximum number
- of open HTTP sessions is reached, legitimate requests are blocked.
-
- Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the
- following issue:
-
-
* AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
- Shell Access
-
- Manager users can execute arbitrary shell commands with the MixMonitor manager
- action. Asterisk does not require system class authorization for a manager
- user to use the MixMonitor action, so any manager user who is permitted to use
- manager commands can potentially execute shell commands as the user executing
- the Asterisk process.
-
- Additionally, the release of 12.3.1 resolves the following issues:
-
-
* AST-2014-005: Remote Crash in PJSIP Channel Driver\'s Publish/Subscribe
- Framework
-
- A remotely exploitable crash vulnerability exists in the PJSIP channel
- driver\'s pub/sub framework. If an attempt is made to unsubscribe when not
- currently subscribed and the endpoint\'s “sub_min_expiry” is set to zero,
- Asterisk tries to create an expiration timer with zero seconds, which is not
- allowed, so an assertion raised.
-
-
* AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions
-
- When a SIP transaction timeout caused a subscription to be terminated, the
- action taken by Asterisk was guaranteed to deadlock the thread on which SIP
- requests are serviced. Note that this behavior could only happen on
- established subscriptions, meaning that this could only be exploited if an
- attacker bypassed authentication and successfully subscribed to a real
- resource on the Asterisk server.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008,
- which were released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert6
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-005.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-008.pdf
Thu Jun 19 14:00:00 2014 Jeffrey Ollie - 11.10.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.10.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-23547 - [patch] app_queue removing callers from queue
- when reloading (Reported by Italo Rossi)
-
* ASTERISK-23559 - app_voicemail fails to load after fix to
- dialplan functions (Reported by Corey Farrell)
-
* ASTERISK-22846 - testsuite: masquerade super test fails on all
- branches (still) (Reported by Matt Jordan)
-
* ASTERISK-23545 - Confbridge talker detection settings
- configuration load bug (Reported by John Knott)
-
* ASTERISK-23546 - CB_ADD_LEN does not do what you\'d think
- (Reported by Walter Doekes)
-
* ASTERISK-23620 - Code path in app_stack fails to unlock list
- (Reported by Bradley Watkins)
-
* ASTERISK-23616 - Big memory leak in logger.c (Reported by
- ibercom)
-
* ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS
- (Reported by Sebastian Wiedenroth)
-
* ASTERISK-23550 - Newer sound sets don\'t show up in menuselect
- (Reported by Rusty Newton)
-
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
-
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
- Krzysztof Chmielewski)
-
* ASTERISK-23605 - res_http_websocket: Race condition in shutting
- down websocket causes crash (Reported by Matt Jordan)
-
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
- PGSQL database state and Asterisk state (Reported by Mark
- Michelson)
-
* ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial
- \'spy\', if the spied-on channel makes a new call, unable to
- barge. (Reported by Robert Moss)
-
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
- (Reported by Guillaume Maudoux)
-
* ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported
- by Guillaume Maudoux)
-
* ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
- for INVITE/w/replaces pickup (Reported by Walter Doekes)
-
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
- (Reported by Steve Davies)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-23649 - [patch]Support for DTLS retransmission
- (Reported by NITESH BANSAL)
-
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
- available in a CLI command (Reported by Patrick Laimbock)
-
* ASTERISK-23754 - [patch] Use var/lib directory for log file
- configured in asterisk.conf (Reported by Igor Goncharovsky)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0
Sat Jun 7 14:00:00 2014 Fedora Release Engineering - 11.9.0-2.1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild
Thu May 15 14:00:00 2014 Dennis Gilmore - 11.9.0-2
- build against gmime-devel not gmime22-devel
- do not use -m64 on aarch64
Wed Apr 23 14:00:00 2014 Jeffrey Ollie - 11.9.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.9.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
- for V.27 (Reported by Paolo Compagnini)
-
* ASTERISK-23034 - [patch] manager Originate doesn\'t abort on
- failed format_cap allocation (Reported by Corey Farrell)
-
* ASTERISK-23061 - [Patch] \'textsupport\' setting not mentioned in
- sip.conf.sample (Reported by Eugene)
-
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
- minus signs (Reported by Jeremy Lainé)
-
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
- from app_queue are not inserted (Reported by Denis Pantsyrev)
-
* ASTERISK-23027 - [patch] Spelling typo \"transfered\" instead of
- \"transferred\" (Reported by Jeremy Lainé)
-
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
- channel connects (Reported by Michael Cargile)
-
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
- request and request queue may differ - fix for locking (Reported
- by adomjan)
-
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
- media offer due to invalid or unsupported syntax (Reported by
- adomjan)
-
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
- GotoIfTime or ExecIfTime causes segmentation fault (Reported by
- Sebastian Murray-Roberts)
-
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
- exceeded (Reported by pz)
-
* ASTERISK-22662 - Documentation fix? - queues.conf says
- persistentmembers defaults to yes, it appears to lie (Reported
- by Rusty Newton)
-
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
- handle selinux port restrictions (Reported by Corey Farrell)
-
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
- crash/core dump (Reported by James Sharp)
-
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI \'reload\'
- command multiple times on cli_aliases (Reported by Joel Vandal)
-
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
- mapping \"module reload\" command (Reported by Gareth Blades)
-
* ASTERISK-17727 - [patch] TLS doesn\'t get all certificate chain
- (Reported by LN)
-
* ASTERISK-23178 - devicestate.h: device state setting functions
- are documented with the wrong return values (Reported by
- Jonathan Rose)
-
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
- is opposite to what\'s expected (Reported by Leon Roy)
-
* ASTERISK-23098 - [patch]possible null pointer dereference in
- format.c (Reported by Marcello Ceschia)
-
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
- res_parking.so is not loaded, or if res_parking.conf has no
- configuration (Reported by CJ Oster)
-
* ASTERISK-23069 - Custom CDR variable not recorded when set in
- macro called from app_queue (Reported by Bryan Anderson)
-
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
- after attended transfer (Reported by Timo Teräs)
-
* ASTERISK-23261 - [patch]Output mixup in
- ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
-
* ASTERISK-23279 - [patch]Asterisk doesn\'t support the dynamic
- payload change in rtp mapping in the 200 OK response (Reported
- by NITESH BANSAL)
-
* ASTERISK-23255 - UUID included for Redhat, but missing for
- Debian distros in install_prereq script (Reported by Rusty
- Newton)
-
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
- variables for subsequent records (Reported by zvision)
-
* ASTERISK-23141 - Asterisk crashes on Dial(), in
- pbx_find_extension at pbx.c (Reported by Maxim)
-
* ASTERISK-23336 - Asterisk warning \"Don\'t know how to indicate
- condition 33 on ooh323c\" on outgoing calls from H323 to SIP peer
- (Reported by Alexander Semych)
-
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
- to minrate=2400, then res_fax refuse to load (Reported by David
- Brillert)
-
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- - probably introduced in 11.7.0 (Reported by OK)
-
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
- handle_response_invite (Reported by Walter Doekes)
-
* ASTERISK-23406 - [patch]Fix typo in \"sip show peer\" (Reported by
- ibercom)
-
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
- (Reported by Jeremy Lainé)
-
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
- from hold (Reported by Vytis Valentinavičius)
-
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
- cause Asterisk to crash (Reported by Joel Vandal)
-
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
- (Reported by John)
-
* ASTERISK-23383 - Wrong sense test on stat return code causes
- unchanged config check to break with include files. (Reported by
- David Woolley)
-
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
- to yes (Reported by Alexandr Gordeev)
-
* ASTERISK-17523 - Qualify for static realtime peers does not work
- (Reported by Maciej Krajewski)
-
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
- unload_module and do_monitor (Reported by Corey Farrell)
-
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
- chan_sip Session-Timers (Reported by Corey Farrell)
-
* ASTERISK-23340 - Security Vulnerability: stack allocation of
- cookie headers in loop allows for unauthenticated remote denial
- of service attack (Reported by Matt Jordan)
-
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
- leaving Conference (Reported by Benjamin Keith Ford)
-
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter
- function in manager.c (Reported by Etienne Lessard)
-
* ASTERISK-23488 - Logic error in callerid checksum processing
- (Reported by Russ Meyerriecks)
-
* ASTERISK-23461 - Only first user is muted when joining
- confbridge with \'startmuted=yes\' (Reported by Chico Manobela)
-
* ASTERISK-20841 - fromdomain not honored on outbound INVITE
- request (Reported by Kelly Goedert)
-
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
- at astobj2.c:120 (Reported by Jamuel Starkey)
-
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to
- play empty files for numbers divisible by 100 (Reported by
- zvision)
-
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
- (Reported by JoshE)
-
* ASTERISK-23391 - Audit dialplan function usage of channel
- variable (Reported by Corey Farrell)
-
* ASTERISK-23548 - POST to ARI sometimes returns no body on
- success (Reported by Scott Griepentrog)
-
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly
- and gatekeeper is not available (Reported by Dmitry Melekhov)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
- against libfreeradius-client (Reported by Jeremy Lainé)
-
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
- not have a call in progress (Reported by Chris Hillman)
-
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
- function to read the whole available data at first and then wait
- for any fragmented packets (Reported by Thava Iyer)
Tue Mar 11 13:00:00 2014 Jeffrey Ollie - 11.8.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
- releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
- and 12.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
-
* AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
-
- Sending a HTTP request that is handled by Asterisk with a large number of
- Cookie headers could overflow the stack.
-
- Another vulnerability along similar lines is any HTTP request with a
- ridiculous number of headers in the request could exhaust system memory.
-
-
* AST-2014-002: chan_sip: Exit early on bad session timers request
-
- This change allows chan_sip to avoid creation of the channel and
- consumption of associated file descriptors altogether if the inbound
- request is going to be rejected anyway.
-
- Additionally, the release of 12.1.1 resolves the following issue:
-
-
* AST-2014-003: res_pjsip: When handling 401/407 responses don\'t assume a
- request will have an endpoint.
-
- This change removes the assumption that an outgoing request will always
- have an endpoint and makes the authenticate_qualify option work once again.
-
- Finally, a security advisory, AST-2014-004, was released for a vulnerability
- fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to
- 12.1.1 to resolve both vulnerabilities.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004,
- which were released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-003.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2014-004.pdf
Tue Mar 4 13:00:00 2014 Jeffrey Ollie - 11.8.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.8.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
* ASTERISK-22544 - Italian prompt vm-options has advertisement in
- it (Reported by Rusty Newton)
-
* ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from
- Asterisk to Chrome (Reported by Shaun Clark)
-
* ASTERISK-22478 - [patch]Can\'t use pound(hash) symbol for custom
- DTMF menus in ConfBridge (processed as directive) (Reported by
- Nicolas Tanski)
-
* ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
- every register message (Reported by Pawel Pierscionek)
-
* ASTERISK-20862 - Asterisk min and max member penalties not
- honored when set with 0 (Reported by Schmooze Com)
-
* ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
- read (Reported by Michael Walton)
-
* ASTERISK-22788 - [patch] main/translate.c: access to variable f
- after free in ast_translate() (Reported by Corey Farrell)
-
* ASTERISK-21242 - Segfault when T.38 re-invite retransmission
- receives 200 OK (Reported by Ashley Winters)
-
* ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
- 16 bit multipart SMS with app_sms (Reported by Jan Juergens)
-
* ASTERISK-22905 - Prevent Asterisk functions that are \'dangerous\'
- from being executed from external interfaces (Reported by Matt
- Jordan)
-
* ASTERISK-23021 - Typos in code : \"avaliable\" instead of
- \"available\" (Reported by Jeremy Lainé)
-
* ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
- by Gareth Palmer)
-
* ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry
- Melekhov)
-
* ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in
- sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger
- \"WIMPy\" Harzenetter)
-
* ASTERISK-22942 - [patch] - Asterisk crashed after
- Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
-
* ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
- instead of seconds (Reported by Robert Mordec)
-
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
- core_event_dispatcher taskprocessor thread (Reported by Etienne
- Lessard)
-
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
- memory when is empty (Reported by Gareth Palmer)
-
* ASTERISK-22871 - cel_pgsql module not loading after \"reload\" or
- \"reload cel_pgsql.so\" command (Reported by Matteo)
-
* ASTERISK-23084 - [patch]rasterisk needlessly prints the
- AST-2013-007 warning (Reported by Tzafrir Cohen)
-
* ASTERISK-17138 - [patch] Asterisk not re-registering after it
- receives \"Forbidden - wrong password on authentication\"
- (Reported by Rudi)
-
* ASTERISK-23011 - [patch]configure.ac and pbx_lua don\'t support
- lua 5.2 (Reported by George Joseph)
-
* ASTERISK-22834 - Parking by blind transfer when lot full orphans
- channels (Reported by rsw686)
-
* ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
- SIP transfer to parking space (Reported by Tommy Thompson)
-
* ASTERISK-22946 - Local From tag regression with sipgate.de
- (Reported by Stephan Eisvogel)
-
* ASTERISK-23010 - No BYE message sent when sip INVITE is received
- (Reported by Ryan Tilton)
-
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- - probably introduced in 11.7.0 (Reported by OK)
-
- Improvements made in this release:
- -----------------------------------
-
* ASTERISK-22728 - [patch] Improve Understanding Of \'Forcerport\'
- When Running \"sip show peers\" (Reported by Michael L. Young)
-
* ASTERISK-22659 - Make a new core and extra sounds release
- (Reported by Rusty Newton)
-
* ASTERISK-22919 - core show channeltypes slicing (Reported by
- outtolunc)
-
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
- output (Reported by outtolunc)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0
Sat Dec 28 13:00:00 2013 Jeffrey Ollie - 11.7.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- app_confbridge: Can now set the language used for announcements
- to the conference.
- (Closes issue ASTERISK-19983. Reported by Jonathan White)
-
-
* --- app_queue: Fix CLI \"queue remove member\" queue_log entry.
- (Closes issue ASTERISK-21826. Reported by Oscar Esteve)
-
-
* --- chan_sip: Do not increment the SDP version between 183 and 200
- responses.
- (Closes issue ASTERISK-21204. Reported by NITESH BANSAL)
-
-
* --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls
- (Closes issue ASTERISK-22005. Reported by Torrey Searle)
-
-
* --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
- And Expires Header In 200ok
- (Closes issue ASTERISK-22428. Reported by Ben Smithurst)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0
Sat Dec 28 13:00:00 2013 Jeffrey Ollie - 11.6.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
- releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4,
- 10.12.4-digiumphones, and 11.6.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
-
* A buffer overflow when receiving odd length 16 bit messages in app_sms. An
- infinite loop could occur which would overwrite memory when a message is
- received into the unpacksms16() function and the length of the message is an
- odd number of bytes.
-
-
* Prevent permissions escalation in the Asterisk Manager Interface. Asterisk
- now marks certain individual dialplan functions as \'dangerous\', which will
- inhibit their execution from external sources.
-
- A \'dangerous\' function is one which results in a privilege escalation. For
- example, if one were to read the channel variable SHELL(rm -rf /) Bad
- Things(TM) could happen; even if the external source has only read
- permissions.
-
- Execution from external sources may be enabled by setting \'live_dangerously\'
- to \'yes\' in the [options] section of asterisk.conf. Although doing so is not
- recommended.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2013-006 and AST-2013-007, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2013-006.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2013-007.pdf
Sat Dec 28 13:00:00 2013 Jeffrey Ollie - 11.6.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Confbridge: empty conference not being torn down
- (Closes issue ASTERISK-21859. Reported by Chris Gentle)
-
-
* --- Let Queue wrap up time influence member availability
- (Closes issue ASTERISK-22189. Reported by Tony Lewis)
-
-
* --- Fix a longstanding issue with MFC-R2 configuration that
- prevented users
- (Closes issue ASTERISK-21117. Reported by Rafael Angulo)
-
-
* --- chan_iax2: Fix saving the wrong expiry time in astdb.
- (Closes issue ASTERISK-22504. Reported by Stefan Wachtler)
-
-
* --- Fix segfault for certain invalid WebSocket input.
- (Closes issue ASTERISK-21825. Reported by Alfred Farrugia)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0
Mon Oct 21 14:00:00 2013 Jeffrey Ollie - 11.5.1-3:
- Disable hardened build, as it\'s apparently causing problems loading modules.
Thu Aug 29 14:00:00 2013 Jeffrey Ollie - 11.5.1-2:
- Enable hardened build BZ#954338
- Significant clean ups
Thu Aug 29 14:00:00 2013 Jeffrey Ollie - 11.5.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases
- are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones,
- and 11.5.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
-
* A remotely exploitable crash vulnerability exists in the SIP channel driver if
- an ACK with SDP is received after the channel has been terminated. The
- handling code incorrectly assumes that the channel will always be present.
-
-
* A remotely exploitable crash vulnerability exists in the SIP channel driver if
- an invalid SDP is sent in a SIP request that defines media descriptions before
- connection information. The handling code incorrectly attempts to reference
- the socket address information even though that information has not yet been
- set.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2013-004 and AST-2013-005, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3-digiumphones
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2013-004.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2013-005.pdf
-
- The Asterisk Development Team has announced the release of Asterisk 11.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fix Segfault In app_queue When \"persistentmembers\" Is Enabled
- And Using Realtime
- (Closes issue ASTERISK-21738. Reported by JoshE)
-
-
* --- IAX2: fix race condition with nativebridge transfers.
- (Closes issue ASTERISK-21409. Reported by alecdavis)
-
-
* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
- Bit
- (Closes issue ASTERISK-21246. Reported by Peter Katzmann)
-
-
* --- Fix One-Way Audio With auto_
* NAT Settings When SIP Calls
- Initiated By PBX
- (Closes issue ASTERISK-21374. Reported by Michael L. Young)
-
-
* --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent
- out after retries fail
- (Closes issue ASTERISK-21677. Reported by Dan Martens)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0
Sat Aug 3 14:00:00 2013 Fedora Release Engineering - 11.4.0-2.2
- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild
Wed Jul 17 14:00:00 2013 Petr Pisar - 11.4.0-2.1
- Perl 5.18 rebuild
Fri May 24 14:00:00 2013 Rex Dieter 11.4.0-2
- rebuild (libical)
Mon May 20 14:00:00 2013 Jeffrey Ollie - 11.4.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fix Sorting Order For Parking Lots Stored In Static Realtime
- (Closes issue ASTERISK-21035. Reported by Alex Epshteyn)
-
-
* --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On
- A Channel
- (Closes issue ASTERISK-21294. Reported by daroz)
-
-
* --- When a session timer expires during a T.38 call, re-invite with
- correct SDP
- (Closes issue ASTERISK-21232. Reported by Nitesh Bansal)
-
-
* --- Fix white noise on SRTP decryption
- (Closes issue ASTERISK-21323. Reported by andrea)
-
-
* --- Fix reload skinny with active devices.
- (Closes issue ASTERISK-16610. Reported by wedhorn)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0
Fri May 10 14:00:00 2013 Tom Callaway - 11.3.0-2:
- fix build with lua 5.2
Tue Apr 23 14:00:00 2013 Jeffrey Ollie - 11.3.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fix issue where chan_mobile fails to bind to first available
- port
- (Closes issue ASTERISK-16357. Reported by challado)
-
-
* --- Fix Queue Log Reporting Every Call COMPLETECALLER With \"h\"
- Extension Present
- (Closes issue ASTERISK-20743. Reported by call)
-
-
* --- Retain XMPP filters across reconnections so external modules
- continue to function as expected.
- (Closes issue ASTERISK-20916. Reported by kuj)
-
-
* --- Ensure that a declined media stream is terminated with a \'\\r\
\'
- (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis)
-
-
* --- Fix pjproject compilation in certain circumstances
- (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0
Thu Mar 28 13:00:00 2013 Jeffrey Ollie - 11.2.2-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
- are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
- and 11.2.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following issues:
-
-
* A possible buffer overflow during H.264 format negotiation. The format
- attribute resource for H.264 video performs an unsafe read against a media
- attribute when parsing the SDP.
-
- This vulnerability only affected Asterisk 11.
-
-
* A denial of service exists in Asterisk\'s HTTP server. AST-2012-014, fixed
- in January of this year, contained a fix for Asterisk\'s HTTP server for a
- remotely-triggered crash. While the fix prevented the crash from being
- triggered, a denial of service vector still exists with that solution if an
- attacker sends one or more HTTP POST requests with very large Content-Length
- values.
-
- This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
-
-
* A potential username disclosure exists in the SIP channel driver. When
- authenticating a SIP request with alwaysauthreject enabled, allowguest
- disabled, and autocreatepeer disabled, Asterisk discloses whether a user
- exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
-
- This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
Sun Feb 10 13:00:00 2013 Jeffrey Ollie - 11.2.1-1:
- The Asterisk Development Team has announced the release of Asterisk 11.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
-
* --- Fix astcanary startup problem due to wrong pid value from before
- daemon call
- (Closes issue ASTERISK-20947. Reported by Jakob Hirsch)
-
-
* --- Update init.d scripts to handle stderr; readd splash screen for
- remote consoles
- (Closes issue ASTERISK-20945. Reported by Warren Selby)
-
-
* --- Reset RTP timestamp; sequence number on SSRC change
- (Closes issue ASTERISK-20906. Reported by Eelco Brolman)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1
Fri Jan 18 13:00:00 2013 Jeffrey Ollie - 11.2.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- app_meetme: Fix channels lingering when hung up under certain
- conditions
- (Closes issue ASTERISK-20486. Reported by Michael Cargile)
-
-
* --- Fix stuck DTMF when bridge is broken.
- (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy)
-
-
* --- Add missing support for \"who hung up\" to chan_motif.
- (Closes issue ASTERISK-20671. Reported by Matt Jordan)
-
-
* --- Remove a fixed size limitation for producing SDP and change how
- ICE support is disabled by default.
- (Closes issue ASTERISK-20643. Reported by coopvr)
-
-
* --- Fix chan_sip websocket payload handling
- (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo)
-
-
* --- Fix pjproject compilation in certain circumstances
- (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0
Thu Jan 3 13:00:00 2013 Jeffrey Ollie - 11.1.2-1:
- The Asterisk Development Team has announced a security release for Asterisk 11,
- Asterisk 11.1.2. This release addresses the security vulnerabilities reported in
- AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11
- released for these security vulnerabilities. The prior release left open a
- vulnerability in res_xmpp that exists only in Asterisk 11; as such, other
- versions of Asterisk were resolved correctly by the previous releases.
-
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following two issues:
-
-
* Stack overflows that occur in some portions of Asterisk that manage a TCP
- connection. In SIP, this is exploitable via a remote unauthenticated session;
- in XMPP and HTTP connections, this is exploitable via remote authenticated
- sessions. The vulnerabilities in SIP and HTTP were corrected in a prior
- release of Asterisk; the vulnerability in XMPP is resolved in this release.
-
-
* A denial of service vulnerability through exploitation of the device state
- cache. Anonymous calls had the capability to create devices in Asterisk that
- would never be disposed of. Handling the cachability of device states
- aggregated via XMPP is handled in this release.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-014 and AST-2012-015.
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
-
- Thank you for your continued support of Asterisk - and we apologize for having
- to do this twice!
Wed Jan 2 13:00:00 2013 Jeffrey Ollie - 11.1.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
- are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
- and 11.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolve the following two issues:
-
-
* Stack overflows that occur in some portions of Asterisk that manage a TCP
- connection. In SIP, this is exploitable via a remote unauthenticated session;
- in XMPP and HTTP connections, this is exploitable via remote authenticated
- sessions.
-
-
* A denial of service vulnerability through exploitation of the device state
- cache. Anonymous calls had the capability to create devices in Asterisk that
- would never be disposed of.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-014 and AST-2012-015, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
Wed Dec 12 13:00:00 2012 Jeffrey Ollie - 11.1.0-1:
- The Asterisk Development Team has announced the release of Asterisk 11.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fix execution of \'i\' extension due to uninitialized variable.
- (Closes issue ASTERISK-20455. Reported by Richard Miller)
-
-
* --- Prevent resetting of NATted realtime peer address on reload.
- (Closes issue ASTERISK-18203. Reported by daren ferreira)
-
-
* --- Fix ConfBridge crash if no timing module loaded.
- (Closes issue ASTERISK-19448. Reported by feyfre)
-
-
* --- Fix the Park \'r\' option when a channel parks itself.
- (Closes issue ASTERISK-19382. Reported by James Stocks)
-
-
* --- Fix an issue where outgoing calls would fail to establish audio
- due to ICE negotiation failures.
- (Closes issue ASTERISK-20554. Reported by mmichelson)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0
Fri Dec 7 13:00:00 2012 Jeffrey Ollie - 11.0.2-1:
- The Asterisk Development Team has announced the release of Asterisk 11.0.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.0.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
-
* --- chan_local: Fix local_pvt ref leak in local_devicestate().
- (Closes issue ASTERISK-20769. Reported by rmudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.2
Wed Dec 5 13:00:00 2012 Dan Horák - 11.0.1-3
- simplify LDFLAGS setting
Fri Nov 30 13:00:00 2012 Dennis Gilmore - 11.0.1-2
- clean up things to allow building on arm arches
Mon Nov 5 13:00:00 2012 Jeffrey Ollie - 11.0.1-1
- The Asterisk Development Team has announced the release of Asterisk 11.0.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.0.1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
-
* --- chan_sip: Fix a bug causing SIP reloads to remove all entries
- from the registry
- (Closes issue ASTERISK-20611. Reported by Alisher)
-
-
* --- confbridge: Fix a bug which made conferences not record with
- AMI/CLI commands
- (Closes issue ASTERISK-20601. Reported by Vilius)
-
-
* --- Fix an issue with res_http_websocket where the chan_sip
- WebSocket handler could not be registered.
- (Closes issue ASTERISK-20631. Reported by danjenkins)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1
Tue Oct 30 13:00:00 2012 Jeffrey Ollie - 11.0.0-1:
- The Asterisk Development Team is pleased to announce the release of
- Asterisk 11.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- Asterisk 11 is the next major release series of Asterisk. It is a Long Term
- Support (LTS) release, similar to Asterisk 1.8. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- For important information regarding upgrading to Asterisk 11, please see the
- Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
-
- A short list of new features includes:
-
-
* A new channel driver named chan_motif has been added which provides support
- for Google Talk and Jingle in a single channel driver. This new channel
- driver includes support for both audio and video, RFC2833 DTMF, all codecs
- supported by Asterisk, hold, unhold, and ringing notification. It is also
- compliant with the current Jingle specification, current Google Jingle
- specification, and the original Google Talk protocol.
-
-
* Support for the WebSocket transport for chan_sip.
-
-
* SIP peers can now be configured to support negotiation of ICE candidates.
-
-
* The app_page application now no longer depends on DAHDI or app_meetme. It
- has been re-architected to use app_confbridge internally.
-
-
* Hangup handlers can be attached to channels using the CHANNEL() function.
- Hangup handlers will run when the channel is hung up similar to the h
- extension; however, unlike an h extension, a hangup handler is associated with
- the actual channel and will execute anytime that channel is hung up,
- regardless of where it is in the dialplan.
-
-
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial
- allows you to execute a dialplan subroutine on a channel before a call is
- placed but after the application performing a dial action is invoked. This
- means that the handlers are executed after the creation of the callee
- channels, but before any actions have been taken to actually dial the callee
- channels.
-
-
* Log messages can now be easily associated with a certain call by looking at
- a new unique identifier, \"Call Id\". Call ids are attached to log messages for
- just about any case where it can be determined that the message is related
- to a particular call.
-
-
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
- Asterisk. Unlike traditional ACLs defined in specific module configuration
- files, Named ACLs can be shared across multiple modules.
-
-
* The Hangup Cause family of functions and dialplan applications allow for
- inspection of the hangup cause codes for each channel involved in a call.
- This allows a dialplan writer to determine, for each channel, who hung up and
- for what reason(s).
-
-
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
- lets you set some of the configuration options from the general section
- of features.conf on a per-channel basis. FEATUREMAP() lets you customize
- the key sequence used to activate built-in features, such as blindxfer,
- and automon.
-
-
* Support for DTLS-SRTP in chan_sip.
-
-
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
- and callgroups to be defined for several channel drivers.
-
-
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
-
- More information about the new features can be found on the Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
-
- A full list of all new features can also be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog.
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0
Wed Oct 17 14:00:00 2012 Jeffrey Ollie - 11.0.0-0.7.rc2:
- The Asterisk Development Team has announced the second release candidate of
- Asterisk 11.0.0. This release candidate is available for immediate
- download at http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.0.0-rc2 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release candidate:
-
-
* --- Fix an issue where outgoing calls would fail to establish audio
- due to ICE negotiation failures.
- (Closes issue ASTERISK-20554. Reported by mmichelson)
-
-
* --- Ensure Asterisk fails TCP/TLS SIP calls when certificate
- checking fails
- (Closes issue ASTERISK-20559. Reported by kmoore)
-
-
* --- Don\'t make chan_sip export global symbols.
- (Closes issue ASTERISK-20545. Reported by kmoore)
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2
Tue Oct 9 14:00:00 2012 Jeffrey Ollie - 11.0.0-0.6.rc1
- The Asterisk Development Team is pleased to announce the first release candidate
- of Asterisk 11.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- All interested users of Asterisk are encouraged to participate in the
- Asterisk 11 testing process. Please report any issues found to the issue
- tracker, https://issues.asterisk.org/jira. It is also very useful to see
- successful test reports. Please post those to the asterisk-dev mailing list.
- All Asterisk users are invited to participate in the #asterisk-testing channel
- on IRC to work together in testing the many parts of Asterisk.
-
- Asterisk 11 is the next major release series of Asterisk. It will be a Long
- Term Support (LTS) release, similar to Asterisk 1.8. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- For important information regarding upgrading to Asterisk 11, please see the
- Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
-
- A short list of new features includes:
-
-
* A new channel driver named chan_motif has been added which provides support
- for Google Talk and Jingle in a single channel driver. This new channel
- driver includes support for both audio and video, RFC2833 DTMF, all codecs
- supported by Asterisk, hold, unhold, and ringing notification. It is also
- compliant with the current Jingle specification, current Google Jingle
- specification, and the original Google Talk protocol.
-
-
* Support for the WebSocket transport for chan_sip.
-
-
* SIP peers can now be configured to support negotiation of ICE candidates.
-
-
* The app_page application now no longer depends on DAHDI or app_meetme. It
- has been re-architected to use app_confbridge internally.
-
-
* Hangup handlers can be attached to channels using the CHANNEL() function.
- Hangup handlers will run when the channel is hung up similar to the h
- extension; however, unlike an h extension, a hangup handler is associated with
- the actual channel and will execute anytime that channel is hung up,
- regardless of where it is in the dialplan.
-
-
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial
- allows you to execute a dialplan subroutine on a channel before a call is
- placed but after the application performing a dial action is invoked. This
- means that the handlers are executed after the creation of the callee
- channels, but before any actions have been taken to actually dial the callee
- channels.
-
-
* Log messages can now be easily associated with a certain call by looking at
- a new unique identifier, \"Call Id\". Call ids are attached to log messages for
- just about any case where it can be determined that the message is related
- to a particular call.
-
-
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
- Asterisk. Unlike traditional ACLs defined in specific module configuration
- files, Named ACLs can be shared across multiple modules.
-
-
* The Hangup Cause family of functions and dialplan applications allow for
- inspection of the hangup cause codes for each channel involved in a call.
- This allows a dialplan writer to determine, for each channel, who hung up and
- for what reason(s).
-
-
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
- lets you set some of the configuration options from the general section
- of features.conf on a per-channel basis. FEATUREMAP() lets you customize
- the key sequence used to activate built-in features, such as blindxfer,
- and automon.
-
-
* Support for DTLS-SRTP in chan_sip.
-
-
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
- and callgroups to be defined for several channel drivers.
-
-
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
-
- More information about the new features can be found on the Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
-
- A full list of all new features can also be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog.
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1
Wed Sep 26 14:00:00 2012 Jeffrey Ollie - 11.0.0-0.5.beta2
- Don\'t forget format_ilbc module
Wed Sep 26 14:00:00 2012 Jeffrey Ollie - 11.0.0-0.4.beta2
- The Asterisk Development Team is pleased to announce the second beta release of
- Asterisk 11.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- All interested users of Asterisk are encouraged to participate in the
- Asterisk 11 testing process. Please report any issues found to the issue
- tracker, https://issues.asterisk.org/jira. It is also very useful to see
- successful test reports. Please post those to the asterisk-dev mailing list.
- All Asterisk users are invited to participate in the #asterisk-testing channel
- on IRC to work together in testing the many parts of Asterisk.
-
- Asterisk 11 is the next major release series of Asterisk. It will be a Long
- Term Support (LTS) release, similar to Asterisk 1.8. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- For important information regarding upgrading to Asterisk 11, please see the
- Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
-
- A short list of new features includes:
-
-
* A new channel driver named chan_motif has been added which provides support
- for Google Talk and Jingle in a single channel driver. This new channel
- driver includes support for both audio and video, RFC2833 DTMF, all codecs
- supported by Asterisk, hold, unhold, and ringing notification. It is also
- compliant with the current Jingle specification, current Google Jingle
- specification, and the original Google Talk protocol.
-
-
* Support for the WebSocket transport for chan_sip.
-
-
* SIP peers can now be configured to support negotiation of ICE candidates.
-
-
* The app_page application now no longer depends on DAHDI or app_meetme. It
- has been re-architected to use app_confbridge internally.
-
-
* Hangup handlers can be attached to channels using the CHANNEL() function.
- Hangup handlers will run when the channel is hung up similar to the h
- extension; however, unlike an h extension, a hangup handler is associated with
- the actual channel and will execute anytime that channel is hung up,
- regardless of where it is in the dialplan.
-
-
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial
- allows you to execute a dialplan subroutine on a channel before a call is
- placed but after the application performing a dial action is invoked. This
- means that the handlers are executed after the creation of the callee
- channels, but before any actions have been taken to actually dial the callee
- channels.
-
-
* Log messages can now be easily associated with a certain call by looking at
- a new unique identifier, \"Call Id\". Call ids are attached to log messages for
- just about any case where it can be determined that the message is related
- to a particular call.
-
-
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
- Asterisk. Unlike traditional ACLs defined in specific module configuration
- files, Named ACLs can be shared across multiple modules.
-
-
* The Hangup Cause family of functions and dialplan applications allow for
- inspection of the hangup cause codes for each channel involved in a call.
- This allows a dialplan writer to determine, for each channel, who hung up and
- for what reason(s).
-
-
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
- lets you set some of the configuration options from the general section
- of features.conf on a per-channel basis. FEATUREMAP() lets you customize
- the key sequence used to activate built-in features, such as blindxfer,
- and automon.
-
-
* Support for DTLS-SRTP in chan_sip.
-
-
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
- and callgroups to be defined for several channel drivers.
-
-
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
-
- More information about the new features can be found on the Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
-
- A full list of all new features can also be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog.
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2
Wed Sep 26 14:00:00 2012 Jeffrey Ollie - 10.8.0-1
- The Asterisk Development Team has announced the release of Asterisk 10.8.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through
- ExternalIVR
- (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research)
-
-
* --- AST-2012-013: Resolve ACL rules being ignored during calls by
- some IAX2 peers
- (Closes issue ASTERISK-20186. Reported by Alan Frisch)
-
-
* --- Handle extremely out of order RFC 2833 DTMF
- (Closes issue ASTERISK-18404. Reported by Stephane Chazelas)
-
-
* --- Resolve severe memory leak in CEL logging modules.
- (Closes issue AST-916. Reported by Thomas Arimont)
-
-
* --- Only re-create an SRTP session when needed
- (Issue ASTERISK-20194. Reported by Nicolo Mazzon)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0
Tue Sep 4 14:00:00 2012 Dan Horák - 11.0.0-0.3.beta1
- fix build on s390
Tue Sep 4 14:00:00 2012 Dan Horák - 10.7.1-2
- fix build on s390
Thu Aug 30 14:00:00 2012 Jeffrey Ollie - 10.7.1-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
- released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones
- resolve the following two issues:
-
-
* A permission escalation vulnerability in Asterisk Manager Interface. This
- would potentially allow remote authenticated users the ability to execute
- commands on the system shell with the privileges of the user running the
- Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt
- file delivered with Asterisk has been updated due to this and other related
- vulnerabilities fixed in previous versions of Asterisk.
-
-
* When an IAX2 call is made using the credentials of a peer defined in a
- dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that
- peer are not applied to the call attempt. This allows for a remote attacker
- who is aware of a peer\'s credentials to bypass the ACL rules set for that
- peer.
-
- These issues and their resolutions are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-012 and AST-2012-013, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert7
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-012.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-013.pdf
Thu Aug 30 14:00:00 2012 Jeffrey Ollie - 10.7.0-1
- The Asterisk Development Team has announced the release of Asterisk 10.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fix deadlock potential with ast_set_hangupsource() calls.
- (Closes issue ASTERISK-19801. Reported by Alec Davis)
-
-
* --- Fix request routing issue when outboundproxy is used.
- (Closes issue ASTERISK-20008. Reported by Marcus Hunger)
-
-
* --- Set the Caller ID \"tag\" on peers even if remote party
- information is present.
- (Closes issue ASTERISK-19859. Reported by Thomas Arimont)
-
-
* --- Fix NULL pointer segfault in ast_sockaddr_parse()
- (Closes issue ASTERISK-20006. Reported by Michael L. Young)
-
-
* --- Do not perform install on existing directories
- (Closes issue ASTERISK-19492. Reported by Karl Fife)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0
Thu Aug 30 14:00:00 2012 Jeffrey Ollie - 10.6.1-1
- The Asterisk Development Team has announced the release of Asterisk 10.6.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.6.1 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
-
* --- Remove a superfluous and dangerous freeing of an SSL_CTX.
- (Closes issue ASTERISK-20074. Reported by Trevor Helmsley)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1
Thu Aug 30 14:00:00 2012 Jeffrey Ollie - 10.6.0-1
- The Asterisk Development Team has announced the release of Asterisk 10.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- format_mp3: Fix a possible crash in mp3_read().
- (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)
-
-
* --- Fix local channel chains optimizing themselves out of a call.
- (Closes issue ASTERISK-16711. Reported by Alec Davis)
-
-
* --- Re-add LastMsgsSent value for SIP peers
- (Closes issue ASTERISK-17866. Reported by Steve Davies)
-
-
* --- Prevent sip_pvt refleak when an ast_channel outlasts its
- corresponding sip_pvt.
- (Closes issue ASTERISK-19425. Reported by David Cunningham)
-
-
* --- Send more accurate identification information in dialog-info SIP
- NOTIFYs.
- (Closes issue ASTERISK-16735. Reported by Maciej Krajewski)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0
Sat Aug 18 14:00:00 2012 Jeffrey Ollie - 11.0.0-0.2.beta1
- The Asterisk Development Team is pleased to announce the first beta release of
- Asterisk 11.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- All interested users of Asterisk are encouraged to participate in the
- Asterisk 11 testing process. Please report any issues found to the issue
- tracker, https://issues.asterisk.org/jira. It is also very useful to see
- successful test reports. Please post those to the asterisk-dev mailing list.
- All Asterisk users are invited to participate in the #asterisk-testing channel
- on IRC to work together in testing the many parts of Asterisk.
-
- Asterisk 11 is the next major release series of Asterisk. It will be a Long
- Term Support (LTS) release, similar to Asterisk 1.8. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- For important information regarding upgrading to Asterisk 11, please see the
- Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
-
- A short list of new features includes:
-
-
* A new channel driver named chan_motif has been added which provides support
- for Google Talk and Jingle in a single channel driver. This new channel
- driver includes support for both audio and video, RFC2833 DTMF, all codecs
- supported by Asterisk, hold, unhold, and ringing notification. It is also
- compliant with the current Jingle specification, current Google Jingle
- specification, and the original Google Talk protocol.
-
-
* Support for the WebSocket transport for chan_sip.
-
-
* SIP peers can now be configured to support negotiation of ICE candidates.
-
-
* The app_page application now no longer depends on DAHDI or app_meetme. It
- has been re-architected to use app_confbridge internally.
-
-
* Hangup handlers can be attached to channels using the CHANNEL() function.
- Hangup handlers will run when the channel is hung up similar to the h
- extension; however, unlike an h extension, a hangup handler is associated with
- the actual channel and will execute anytime that channel is hung up,
- regardless of where it is in the dialplan.
-
-
* Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial
- allows you to execute a dialplan subroutine on a channel before a call is
- placed but after the application performing a dial action is invoked. This
- means that the handlers are executed after the creation of the caller/callee
- channels, but before any actions have been taken to actually dial the callee
- channels.
-
-
* Log messages can now be easily associated with a certain call by looking at
- a new unique identifier, \"Call Id\". Call ids are attached to log messages for
- just about any case where it can be determined that the message is related
- to a particular call.
-
-
* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
- Asterisk. Unlike traditional ACLs defined in specific module configuration
- files, Named ACLs can be shared across multiple modules.
-
-
* The Hangup Cause family of functions and dialplan applications allow for
- inspection of the hangup cause codes for each channel involved in a call.
- This allows a dialplan writer to determine, for each channel, who hung up and
- for what reason(s).
-
-
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
- lets you set some of the configuration options from the general section
- of features.conf on a per-channel basis. FEATUREMAP() lets you customize
- the key sequence used to activate built-in features, such as blindxfer,
- and automon.
-
-
* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
- and callgroups to be defined for several channel drivers.
-
-
* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
-
- More information about the new features can be found on the Asterisk wiki:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
-
- A full list of all new features can also be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog.
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1
Wed Jul 18 14:00:00 2012 Fedora Release Engineering - 10.5.2-1.2
- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild
Mon Jul 9 14:00:00 2012 Petr Pisar - 10.5.2-1.1
- Perl 5.16 rebuild
Thu Jul 5 14:00:00 2012 Jeffrey Ollie - 10.5.2-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
- released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones
- resolve the following two issues:
-
-
* If Asterisk sends a re-invite and an endpoint responds to the re-invite with
- a provisional response but never sends a final response, then the SIP dialog
- structure is never freed and the RTP ports for the call are never released. If
- an attacker has the ability to place a call, they could create a denial of
- service by using all available RTP ports.
-
-
* If a single voicemail account is manipulated by two parties simultaneously,
- a condition can occur where memory is freed twice causing a crash.
-
- These issues and their resolution are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-010 and AST-2012-011, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-010.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-011.pdf
Thu Jun 28 14:00:00 2012 Petr Pisar - 10.5.1-1.1
- Perl 5.16 rebuild
Fri Jun 15 14:00:00 2012 Jeffrey Ollie - 10.5.1-1
- The Asterisk Development Team has announced a security release for Asterisk 10.
- This security release is released as version 10.5.1.
-
- The release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 10.5.1 resolves the following issue:
-
-
* A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
- Channel driver. When an SCCP client sends an Off Hook message, followed by
- a Key Pad Button Message, a structure that was previously set to NULL is
- dereferenced. This allows remote authenticated connections the ability to
- cause a crash in the server, denying services to legitimate users.
-
- This issue and its resolution is described in the security advisory.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2012-009, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.1
-
- The security advisory is available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-009.pdf
Fri Jun 15 14:00:00 2012 Jeffrey Ollie - 10.5.0-1
- The Asterisk Development Team has announced the release of Asterisk 10.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Turn off warning message when bind address is set to any.
- (Closes issue ASTERISK-19456. Reported by Michael L. Young)
-
-
* --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit
- machines
- (Closes issue ASTERISK-19727. Reported by Ben Klang)
-
-
* --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply
- before disconnecting the call.
- (Closes issue ASTERISK-19708. Reported by mehdi Shirazi)
-
-
* --- Fix recalled party B feature flags for a failed DTMF atxfer.
- (Closes issue ASTERISK-19383. Reported by lgfsantos)
-
-
* --- Fix DTMF atxfer running h exten after the wrong bridge ends.
- (Closes issue ASTERISK-19717. Reported by Mario)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0
Mon Jun 11 14:00:00 2012 Petr Pisar - 10.4.2-1.1
- Perl 5.16 rebuild
Wed May 30 14:00:00 2012 Jeffrey Ollie - 10.4.2-1
- The Asterisk Development Team has announced the release of Asterisk 10.4.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.4.2 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
-
* --- Resolve crash in subscribing for MWI notifications
- (Closes issue ASTERISK-19827. Reported by B. R)
-
-
* --- Fix crash in ConfBridge when user announcement is played for
- more than 2 users
- (Closes issue ASTERISK-19899. Reported by Florian Gilcher)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2
Wed May 30 14:00:00 2012 Jeffrey Ollie - 10.4.1-1
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
- released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following
- two issues:
-
-
* A remotely exploitable crash vulnerability exists in the IAX2 channel
- driver if an established call is placed on hold without a suggested music
- class. Asterisk will attempt to use an invalid pointer to the music
- on hold class name, potentially causing a crash.
-
-
* A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
- Channel driver. When an SCCP client closes its connection to the server,
- a pointer in a structure is set to NULL. If the client was not in the
- on-hook state at the time the connection was closed, this pointer is later
- dereferenced. This allows remote authenticated connections the ability to
- cause a crash in the server, denying services to legitimate users.
-
- These issues and their resolution are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-007 and AST-2012-008, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-007.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-008.pdf
Fri May 4 14:00:00 2012 Jeffrey Ollie - 10.4.0-1
- The Asterisk Development Team has announced the release of Asterisk 10.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 10.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
-
* --- Prevent chanspy from binding to zombie channels
- (Closes issue ASTERISK-19493. Reported by lvl)
-
-
* --- Fix Dial m and r options and forked calls generating warnings
- for voice frames.
- (Closes issue ASTERISK-16901. Reported by Chris Gentle)
-
-
* --- Remove ISDN hold restriction for non-bridged calls.
- (Closes issue ASTERISK-19388. Reported by Birger Harzenetter)
-
-
* --- Fix copying of CDR(accountcode) to local channels.
- (Closes issue ASTERISK-19384. Reported by jamicque)
-
-
* --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
- (Closes issue ASTERISK-19303. Reported by Jon Tsiros)
-
-
* --- Eliminate double close of file descriptor in manager.c
- (Closes issue ASTERISK-18453. Reported by Jaco Kroon)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0
Tue Apr 24 14:00:00 2012 Jeffrey Ollie - 10.3.1-1
- The Asterisk Development Team has announced security releases for Asterisk 1.6.2,
- 1.8, and 10. The available security releases are released as versions 1.6.2.24,
- 1.8.11.1, and 10.3.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the following two
- issues:
-
-
* A permission escalation vulnerability in Asterisk Manager Interface. This
- would potentially allow remote authenticated users the ability to execute
- commands on the system shell with the privileges of the user running the
- Asterisk application.
-
-
* A heap overflow vulnerability in the Skinny Channel driver. The keypad
- button message event failed to check the length of a fixed length buffer
- before appending a received digit to the end of that buffer. A remote
- authenticated user could send sufficient keypad button message events that the
- buffer would be overrun.
-
- In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve the following
- issue:
-
-
* A remote crash vulnerability in the SIP channel driver when processing UPDATE
- requests. If a SIP UPDATE request was received indicating a connected line
- update after a channel was terminated but before the final destruction of the
- associated SIP dialog, Asterisk would attempt a connected line update on a
- non-existing channel, causing a crash.
-
- These issues and their resolution are described in the security advisories.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2012-004, AST-2012-005, and AST-2012-006, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.24
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1
-
- The security advisories are available at:
-
-
* http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-005.pdf
-
* http://downloads.asterisk.org/pub/security/AST-2012-006.pdf
Thu Mar 29 14:00:00 2012 Russell Bryant - 10.3.0-1
- Update to 10.3.0
Fri Mar 16 13:00:00 2012 Russell Bryant - 10.2.1-1
- Update to 10.2.1 from upstream.
- Fix remote stack overflow in app_milliwatt.
- Fix remote stack overflow, including possible code injection, in HTTP digest
authentication handling.
- Disable asterisk-corosync package, as it doesn\'t build right now.
- Resolves: rhbz#804045, rhbz#804038, rhbz#804042
Thu Feb 16 13:00:00 2012 Jeffrey C. Ollie - 10.1.2-2
-
* Add patch extracted from upstream to build with Corosync since
- OpenAIS is no longer available.
-
* Add PrivateTmp=true to systemd service file (#782478)
-
* Add some macros to make it easier to build with fewer dependencies
- (with corresponding less functionality) (#787389)
-
* Add isa macros in a few places plus a few other changes to make it
- easier to cross-compile. (#787779)
Thu Feb 16 13:00:00 2012 Jeffrey C. Ollie - 10.1.2-1
- The Asterisk Development Team has announced the release of Asterisk 10.1.2. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 10.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
-
* --- Fix SIP INFO DTMF handling for non-numeric codes ---
- (Closes issue ASTERISK-19290. Reported by: Ira Emus)
-
-
* --- Fix crash in ParkAndAnnounce ---
- (Closes issue ASTERISK-19311. Reported-by: tootai)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2
Thu Feb 16 13:00:00 2012 Jeffrey C. Ollie - 10.1.1-1
- The Asterisk Development Team has announced the release of Asterisk 10.1.1. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 10.1.1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* --- Fixes deadlocks occuring in chan_agent ---
- (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)
-
-
* --- Ensure entering T.38 passthrough does not cause an infinite loop ---
- (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1
Thu Feb 16 13:00:00 2012 Jeffrey C. Ollie - 10.1.0-1
- The Asterisk Development Team is pleased to announce the release of
- Asterisk 10.1.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 10.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* AST-2012-001: prevent crash when an SDP offer
- is received with an encrypted video stream when support for video
- is disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
- Reported by: Catalin Sanda
-
-
* Allow playback of formats that don\'t support seeking. ast_streamfile
- previously did unconditional seeking on files that broke playback of
- formats that don\'t support that functionality. This patch avoids the
- seek that was causing the problem.
- (closes issue ASTERISK-18994) Patched by: Timo Teras
-
-
* Add pjmedia probation concepts to res_rtp_asterisk\'s learning mode. In
- order to better handle RTP sources with strictrtp enabled (which is the
- default setting in 10) using the learning mode to figure out new sources
- when they change is handled by checking for a number of consecutive (by
- sequence number) packets received to an rtp struct based on a new
- configurable value called \'probation\'. Also, during learning mode instead
- of liberally accepting all packets received, we now reject packets until a
- clear source has been determined.
-
-
* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing
- to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
- causes the loop to exit prematurely. This causes a variety of negative side
- effects, depending on when the loop exits. This patch handles the frame by
- essentially swallowing the frame in the local loop, as the current channel
- drivers expect the RTP bridge to handle the frame, and, in the case of the
- local bridge loop, no additional action is necessary.
- (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
- by: Matt Jordan
-
-
* Fix timing source dependency issues with MOH. Prior to this patch,
- res_musiconhold existed at the same module priority level as the timing
- sources that it depends on. This would cause a problem when music on
- hold was reloaded, as the timing source could be changed after
- res_musiconhold was processed. This patch adds a new module priority
- level, AST_MODPRI_TIMING, that the various timing modules are now loaded
- at. This now occurs before loading other resource modules, such
- that the timing source is guaranteed to be set prior to resolving
- the timing source dependencies.
- (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
- Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
- Patched by elguero
-
-
* Fix RTP reference leak. If a blind transfer were initiated using a
- REFER without a prior reINVITE to place the call on hold, AND if Asterisk
- were sending RTCP reports, then there was a reference leak for the
- RTP instance of the transferrer.
- (closes issue ASTERISK-19192) Reported by: Tyuta Vitali
-
-
* Fix blind transfers from failing if an \'h\' extension
- is present. This prevents the \'h\' extension from being run on the
- transferee channel when it is transferred via a native transfer
- mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported
- by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
- Mark Michelson (license 5049)
-
-
* Restore call progress code for analog ports. Extracting sig_analog
- from chan_dahdi lost call progress detection functionality. Fix
- analog ports from considering a call answered immediately after
- dialing has completed if the callprogress option is enabled.
- (closes issue ASTERISK-18841)
- Reported by: Richard Miller Patched by Richard Miller
-
-
* Fix regression that \'rtp/rtcp set debup ip\' only works when a port
- was also specified.
- (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
- Walter Doekes
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0
Thu Feb 16 13:00:00 2012 Russell Bryant - 10.0.0-2
- Remove asterisk-ais. OpenAIS was removed from Fedora.
Thu Jan 12 13:00:00 2012 Fedora Release Engineering - 10.0.0-1.1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild
Tue Jan 3 13:00:00 2012 Jeffrey C. Ollie - 10.0.0-1
- Don\'t build API docs as the build never finishes
Thu Dec 15 13:00:00 2011 Jeffrey C. Ollie - 10.0.0-1
- The Asterisk Development Team is proud to announce the release of
- Asterisk 10.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- Asterisk 10 is the next major release series of Asterisk. It will be a
- Standard support release, similar to Asterisk 1.6.2. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page:
-
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- With the release of the Asterisk 10 branch, the preceding \'1.\' has been removed
- from the version number per the blog post available at
-
-
- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
-
- The release of Asterisk 10 would not have been possible without the support and
- contributions of the community.
-
- You can find an overview of the work involved with the 10.0.0 release in the
- summary:
-
- http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt
-
- A short list of available features includes:
-
-
* T.38 gateway functionality has been added to res_fax.
-
* Protocol independent out-of-call messaging support. Text messages not
- associated with an active call can now be routed through the Asterisk
- dialplan. SIP and XMPP are supported so far.
-
* New highly optimized and customizable ConfBridge application capable of mixing
- audio at sample rates ranging from 8kHz-192kHz
-
* Addition of video_mode option in confbridge.conf to provide basic video
- conferencing in the ConfBridge() dialplan application.
-
* Support for defining hints has been added to pbx_lua.
-
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES
-
- Also, when upgrading a system between major versions, it is imperative that you
- read and understand the contents of the UPGRADE.txt file, which is located at:
-
- http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt
Fri Dec 9 13:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.7.rc3
- The Asterisk Development Team has announced the third release candidate of
- Asterisk 10.0.0. This release candidate is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 10.0.0-rc3 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release candidate:
-
-
* Add ASTSBINDIR to the list of configurable paths
-
- This patch also makes astdb2sqlite3 and astcanary use the configured
- directory instead of relying on $PATH.
-
-
* Don\'t crash on INFO automon request with no channel
-
- AST-2011-014. When automon was enabled in features.conf, it was possible
- to crash Asterisk by sending an INFO request if no channel had been
- created yet.
-
-
* Fixed crash from orphaned MWI subscriptions in chan_sip
-
- This patch resolves the issue where MWI subscriptions are orphaned
- by subsequent SIP SUBSCRIBE messages.
-
-
* Fix a change in behavior in \'database show\' from 1.8.
-
- In 1.8 and previous versions, one could use any fullword portion of
- the key name, including the full key, to obtain the record. Until this
- patch, this did not work for the full key.
-
-
* Default to nat=yes; warn when nat in general and peer differ
-
- AST-2011-013. It is possible to enumerate SIP usernames when the general and
- user/peer nat settings differ in whether to respond to the port a request is
- sent from or the port listed for responses in the Via header. In 1.4 and
- 1.6.2, this would mean if one setting was nat=yes or nat=route and the other
- was either nat=no or nat=never. In 1.8 and 10, this would mean when one
- was nat=force_rport and the other was nat=no.
-
- In order to address this problem, it was decided to switch the default
- behavior to nat=yes/force_rport as it is the most commonly used option
- and to strongly discourage setting nat per-peer/user when at all
- possible.
-
-
* Fixed SendMessage stripping extension from To: header in SIP MESSAGE
-
- When using the MessageSend application to send a SIP MESSAGE to a
- non-peer, chan_sip stripped off the extension and failed to add it back
- to the sip_pvt structure before transmitting. This patch adds the full
- URI passed in from the message core to the sip_pvt structure.
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3
Wed Nov 16 13:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.6.rc2
- The Asterisk Development Team has announced the second release candidate of
- Asterisk 10.0.0. This release candidate is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 10.0.0-rc2 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release candidate:
-
-
* Ensure that a null vmexten does not cause a segfault
-
-
* Fix issue with ConfBridge participants hanging up during DTMF feature
- menu usage getting stuck in conference forever
- (closes issue ASTERISK-18829)
- Reported by: zvision
-
-
* Fix app_macro.c MODULEINFO section termination
- (closes issue ASTERISK-18848)
- Reported by: Tony Mountifield
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2
Fri Nov 11 13:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.5.rc1
- The Asterisk Development Team is pleased to announce the first release candidate
- of Asterisk 10.0.0. This release candidate is available for immediate download
- at http://downloads.asterisk.org/pub/telephony/asterisk/
-
- All Asterisk users are encouraged to participate in the Asterisk 10 testing
- process. Please report any issues found to the issue tracker,
- https://issues.asterisk.org/jira. It is also very useful to see successful test
- reports. Please post those to the asterisk-dev mailing list.
-
- All Asterisk users are invited to participate in the #asterisk-testing
- channel on IRC to work together in testing the many parts of Asterisk.
-
- Asterisk 10 is the next major release series of Asterisk. It will be a
- Standard support release, similar to Asterisk 1.6.2. For more
- information about support time lines for Asterisk releases, see the Asterisk
- versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- A short list of features includes:
-
-
* T.38 gateway functionality has been added to res_fax.
-
* Protocol independent out-of-call messaging support. Text messages not
- associated with an active call can now be routed through the Asterisk
- dialplan. SIP and XMPP are supported so far.
-
* New highly optimized and customizable ConfBridge application capable of mixing
- audio at sample rates ranging from 8kHz-192kHz
- (More information available at
- https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
-
* Addition of video_mode option in confbridge.conf to provide basic video
- conferencing in the ConfBridge() dialplan application.
-
* Support for defining hints has been added to pbx_lua.
-
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1
Tue Oct 18 14:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.4.beta2
- Add patch from upstream SVN to fix AST-2011-012
Fri Oct 14 14:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.3.beta2
- Patch cleanup day
Thu Sep 29 14:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.2.beta2
- The Asterisk Development Team is pleased to announce the second beta release of
- Asterisk 10.0.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- With the release of the Asterisk 10 branch, the preceding \'1.\' has been removed
- from the version number per the blog post available at
- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
-
- All interested users of Asterisk are encouraged to participate in the
- Asterisk 10 testing process. Please report any issues found to the issue
- tracker, https://issues.asterisk.org/jira. It is also very useful to see
- successful test reports. Please post those to the asterisk-dev mailing list.
-
- All Asterisk users are invited to participate in the #asterisk-testing
- channel on IRC to work together in testing the many parts of Asterisk.
-
- Asterisk 10 is the next major release series of Asterisk. It will be a
- Standard support release, similar to Asterisk 1.6.2. For more
- information about support time lines for Asterisk releases, see the Asterisk
- versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- A short list of features includes:
-
-
* T.38 gateway functionality has been added to res_fax.
-
-
* Protocol independent out-of-call messaging support. Text messages not
- associated with an active call can now be routed through the Asterisk
- dialplan. SIP and XMPP are supported so far.
-
-
* New highly optimized and customizable ConfBridge application capable of mixing
- audio at sample rates ranging from 8kHz-192kHz
-
-
* Addition of video_mode option in confbridge.conf to provide basic video
- conferencing in the ConfBridge() dialplan application.
-
-
* Support for defining hints has been added to pbx_lua.
-
-
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
-
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2
Mon Jul 25 14:00:00 2011 Jeffrey C. Ollie - 10.0.0-0.1.beta1
-
- The Asterisk Development Team is pleased to announce the first beta release of
- Asterisk 10.0.0-beta1. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- With the release of the Asterisk 10 branch, the preceding \'1.\' has been removed
- from the version number per the blog post available at
- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
-
- All interested users of Asterisk are encouraged to participate in the
- Asterisk 10 testing process. Please report any issues found to the issue
- tracker, https://issues.asterisk.org/jira. It is also very useful to see
- successful test reports. Please post those to the asterisk-dev mailing list.
-
- All Asterisk users are invited to participate in the #asterisk-testing
- channel on IRC to work together in testing the many parts of Asterisk.
- Additionally users can make use of the RPM and DEB packages now being built for
- all Asterisk releases. More information available at
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
-
- Asterisk 10 is the next major release series of Asterisk. It will be a
- Standard support release, similar to Asterisk 1.6.2. For more
- information about support time lines for Asterisk releases, see the Asterisk
- versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
-
- A short list of included features includes:
-
-
* T.38 gateway functionality has been added to res_fax.
-
* Protocol independent out-of-call messaging support. Text messages not
- associated with an active call can now be routed through the Asterisk
- dialplan. SIP and XMPP are supported so far.
-
* New highly optimized and customizable ConfBridge application capable of mixing
- audio at sample rates ranging from 8kHz-192kHz
-
* Addition of video_mode option in confbridge.conf to provide basic video
- conferencing in the ConfBridge() dialplan application.
-
* Support for defining hints has been added to pbx_lua.
-
* Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1
Thu Jul 21 14:00:00 2011 Petr Sabata - 1.8.5.0-1.2
- Perl mass rebuild
Wed Jul 20 14:00:00 2011 Petr Sabata - 1.8.5.0-1.1
- Perl mass rebuild
Mon Jul 11 14:00:00 2011 Jeffrey C. Ollie - 1.8.5.0-1
- The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* Fix Deadlock with attended transfer of SIP call
- (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
- cmaj)
-
-
* Fixes thread blocking issue in the sip TCP/TLS implementation.
- (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
- rossbeer, kowalma, Freddi_Fonet)
-
-
* Be more tolerant of what URI we accept for call completion PUBLISH requests.
- (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
-
-
* Fix a nasty chanspy bug which was causing a channel leak every time a spied on
- channel made a call.
- (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
-
-
* This patch fixes a bug with MeetMe behavior where the \'P\' option for always
- prompting for a pin is ignored for the first caller.
- (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
-
-
* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
- the call that the dialplan started an AGI script for is hungup while the AGI
- script is in the middle of a command then the AGI script is not notified of
- the hangup.
- (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
-
-
* Resolve issue where leaving a voicemail, the MWI message is never sent. The
- same thing happens when checking a voicemail and marking it as read.
- (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
- Mudgett)
-
-
* Resolve issue where wait for leader with Music On Hold allows crosstalk
- between participants. Parenthesis in the wrong position. Regression from issue
- #14365 when expanding conference flags to use 64 bits.
- (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0
Thu Jul 7 14:00:00 2011 Jeffrey C. Ollie - 1.8.5-0.2
- Rebuild for net-snmp 5.7
Fri Jul 1 14:00:00 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1
- Fix systemd dependencies in EL6 and F15
Thu Jun 30 14:00:00 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1
- The Asterisk Development Team has announced the first release candidate of
- Asterisk 1.8.5. This release candidate is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release candidate:
-
-
* Fix Deadlock with attended transfer of SIP call
- (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
- cmaj)
-
-
* Fixes thread blocking issue in the sip TCP/TLS implementation.
- (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
- rossbeer, kowalma, Freddi_Fonet)
-
-
* Be more tolerant of what URI we accept for call completion PUBLISH requests.
- (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
-
-
* Fix a nasty chanspy bug which was causing a channel leak every time a spied on
- channel made a call.
- (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
-
-
* This patch fixes a bug with MeetMe behavior where the \'P\' option for always
- prompting for a pin is ignored for the first caller.
- (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
-
-
* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
- the call that the dialplan started an AGI script for is hungup while the AGI
- script is in the middle of a command then the AGI script is not notified of
- the hangup.
- (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
-
-
* Resolve issue where leaving a voicemail, the MWI message is never sent. The
- same thing happens when checking a voicemail and marking it as read.
- (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
- Mudgett)
-
-
* Resolve issue where wait for leader with Music On Hold allows crosstalk
- between participants. Parenthesis in the wrong position. Regression from issue
- #14365 when expanding conference flags to use 64 bits.
- (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
-
-
* Fix timerfd locking issue.
- (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1
Thu Jun 30 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.4-2
- Fedora Directory Server -> 389 Directory Server
Wed Jun 29 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.4-1
- The Asterisk Development Team has announced the release of Asterisk
- versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security
- releases.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the
- following issue:
-
- AST-2011-011: Asterisk may respond differently to SIP requests from an
- invalid SIP user than it does to a user configured on the system, even
- when the alwaysauthreject option is set in the configuration. This can
- leak information about what SIP users are valid on the Asterisk
- system.
-
- For more information about the details of this vulnerability, please
- read the security advisory AST-2011-011, which was released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4
-
- Security advisory AST-2011-011 is available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-011.pdf
Mon Jun 27 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.3-3
- Don\'t forget stereorize
Mon Jun 27 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.3-2
- Move /var/run/asterisk to /run/asterisk
- Add comments to systemd service file on how to mimic safe_asterisk functionality
- Build more of the optional binaries
- Install the tmpfiles.d configuration on Fedora 15
Fri Jun 24 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.3-1
- The Asterisk Development Team has announced the release of Asterisk versions
- 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues
- as outlined below:
-
-
* AST-2011-008: If a remote user sends a SIP packet containing a null,
- Asterisk assumes available data extends past the null to the
- end of the packet when the buffer is actually truncated when
- copied. This causes SIP header parsing to modify data past
- the end of the buffer altering unrelated memory structures.
- This vulnerability does not affect TCP/TLS connections.
- -- Resolved in 1.6.2.18.1 and 1.8.4.3
-
-
* AST-2011-009: A remote user sending a SIP packet containing a Contact header
- with a missing left angle bracket (<) causes Asterisk to
- access a null pointer.
- -- Resolved in 1.8.4.3
-
-
* AST-2011-010: A memory address was inadvertently transmitted over the
- network via IAX2 via an option control frame and the remote party would try
- to access it.
- -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3
-
- The issues and resolutions are described in the AST-2011-008, AST-2011-009, and
- AST-2011-010 security advisories.
-
- For more information about the details of these vulnerabilities, please read
- the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were
- released at the same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3
-
- Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available
- at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-008.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-009.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-010.pdf
Tue Jun 21 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.2-2
- Convert to systemd
Fri Jun 17 14:00:00 2011 Marcela Mašláňová - 1.8.4.2-1.2
- Perl mass rebuild
Fri Jun 10 14:00:00 2011 Marcela Mašláňová - 1.8.4.2-1.1
- Perl 5.14 mass rebuild
Fri Jun 3 14:00:00 2011 Jeffrey C. Ollie - 1.8.4.2-1:
-
- The Asterisk Development Team has announced the release of Asterisk
- version 1.8.4.2, which is a security release for Asterisk 1.8.
-
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of Asterisk 1.8.4.2 resolves an issue with SIP URI
- parsing which can lead to a remotely exploitable crash:
-
- Remote Crash Vulnerability in SIP channel driver (AST-2011-007)
-
- The issue and resolution is described in the AST-2011-007 security
- advisory.
-
- For more information about the details of this vulnerability, please
- read the security advisory AST-2011-007, which was released at the
- same time as this announcement.
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2
-
- Security advisory AST-2011-007 is available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-007.pdf
-
- The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.4.1 resolves several issues reported by the
- community. Without your help this release would not have been possible.
- Thank you!
-
- Below is a list of issues resolved in this release:
-
-
* Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
- (Closes issue #18951. Reported by jmls. Patched by wdoekes)
-
-
* Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
- This issue was found and reported by the Asterisk test suite.
- (Closes issue #18951. Patched by mnicholson)
-
-
* Resolve potential crash when using SIP TLS support.
- (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by
- vois, Chainsaw)
-
-
* Improve reliability when using SIP TLS.
- (Closes issue #19182. Reported by st. Patched by mnicholson)
-
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1
- The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.4 resolves several issues reported by the community.
- Without your help this release would not have been possible. Thank you!
-
- Below is a sample of the issues resolved in this release:
-
-
* Use SSLv23_client_method instead of old SSLv2 only.
- (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
- and chazzam.
-
-
* Resolve crash in ast_mutex_init()
- (Patched by twilson)
-
-
* Resolution of several DTMF based attended transfer issues.
- (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
- shihchuan, grecco. Patched by rmudgett)
-
- NOTE: Be sure to read the ChangeLog for more information about these changes.
-
-
* Resolve deadlocks related to device states in chan_sip
- (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)
-
-
* Resolve an issue with the Asterisk manager interface leaking memory when
- disabled.
- (Reported internally by kmorgan. Patched by russellb)
-
-
* Support greetingsfolder as documented in voicemail.conf.sample.
- (Closes issue #17870. Reported by edhorton. Patched by seanbright)
-
-
* Fix channel redirect out of MeetMe() and other issues with channel softhangup
- (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
- Patched by russellb)
-
-
* Fix voicemail sequencing for file based storage.
- (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
- jpeeler)
-
-
* Set hangup cause in local_hangup so the proper return code of 486 instead of
- 503 when using Local channels when the far sides returns a busy. Also affects
- CCSS in Asterisk 1.8+.
- (Patched by twilson)
-
-
* Fix issues with verbose messages not being output to the console.
- (Closes issue #18580. Reported by pabelanger. Patched by qwell)
-
-
* Fix Deadlock with attended transfer of SIP call
- (Closes issue #18837. Reported, patched by alecdavis. Tested by
- alecdavid, Irontec, ZX81, cmaj)
-
- Includes changes per AST-2011-005 and AST-2011-006
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4
-
- Information about the security releases are available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
Thu Apr 21 14:00:00 2011 Jeffrey C. Ollie - 1.8.3.3-1
- The Asterisk Development Team has announced security releases for Asterisk
- branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
- released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
- issues:
-
-
* File Descriptor Resource Exhaustion (AST-2011-005)
-
* Asterisk Manager User Shell Access (AST-2011-006)
-
- The issues and resolutions are described in the AST-2011-005 and AST-2011-006
- security advisories.
-
- For more information about the details of these vulnerabilities, please read the
- security advisories AST-2011-005 and AST-2011-006, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.40.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3
-
- Security advisory AST-2011-005 and AST-2011-006 are available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
Wed Mar 23 13:00:00 2011 Jeffrey C. Ollie - 1.8.3.2-2
- Bump release and rebuild for mysql 5.5.10 soname change.
Thu Mar 17 13:00:00 2011 Jeffrey C. Ollie - 1.8.3.2-1
- The Asterisk Development Team has announced security releases for Asterisk
- branches 1.6.1, 1.6.2, and 1.8. The available security releases are
- released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
-
*
* This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
- contained a bug which caused duplicate manager entries (issue #18987).
-
- The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:
-
-
* Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
-
* Remote crash vulnerability in TCP/TLS server (AST-2011-004)
-
- The issues and resolutions are described in the AST-2011-003 and AST-2011-004
- security advisories.
-
- For more information about the details of these vulnerabilities, please read the
- security advisories AST-2011-003 and AST-2011-004, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.24
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.2
-
- Security advisory AST-2011-003 and AST-2011-004 are available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
Thu Mar 17 13:00:00 2011 Jeffrey C. Ollie - 1.8.3.1-1
- The Asterisk Development Team has announced security releases for Asterisk
- branches 1.6.1, 1.6.2, and 1.8. The available security releases are
- released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:
-
-
* Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
-
* Remote crash vulnerability in TCP/TLS server (AST-2011-004)
-
- The issues and resolutions are described in the AST-2011-003 and AST-2011-004
- security advisories.
-
- For more information about the details of these vulnerabilities, please read the
- security advisories AST-2011-003 and AST-2011-004, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.23
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1
-
- Security advisory AST-2011-003 and AST-2011-004 are available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
- http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
Mon Feb 28 13:00:00 2011 - 1.8.3-1
- The Asterisk Development Team has announced the release of Asterisk 1.8.3. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.3 resolves several issues reported by the community
- and would have not been possible without your participation. Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* Resolve duplicated data in the AstDB when using DIALGROUP()
- (Closes issue #18091. Reported by bunny. Patched by tilghman)
-
-
* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
- (Closes issue #18464. Reported, patched by IgorG)
-
-
* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
- unit tests for the function that does the parsing.
- (Closes issue #18350. Reported by gbour. Patched by Marquis)
-
-
* When using cdr_pgsql the billsec field was not populated correctly on
- unanswered calls.
- (Closes issue #18406. Reported by joscas. Patched by tilghman)
-
-
* Resolve memory leak in iCalendar and Exchange calendaring modules.
- (Closes issue #18521. Reported, patched by pitel. Tested by cervajs)
-
-
* This version of Asterisk includes the new Compiler Flags option
- BETTER_BACKTRACES which uses libbfd to search for better symbol information
- within both the Asterisk binary, as well as loaded modules, to assist when
- using inline backtraces to track down problems.
- (Patched by tilghman)
-
-
* Resolve issue where no Music On Hold may be triggered when using
- res_timing_dahdi.
- (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
- by francesco_r, rfrantik, one47)
-
-
* Resolve a memory leak when the Asterisk Manager Interface is disabled.
- (Reported internally by kmorgan. Patched by russellb)
-
-
* Reimplemented fax session reservation to reverse the ABI breakage introduced
- in r297486.
- (Reported internally. Patched by mnicholson)
-
-
* Fix regression that changed behavior of queues when ringing a queue member.
- (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
-
-
* Resolve deadlock involving REFER.
- (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
-
- Additionally, this release has the changes related to security bulletin
- AST-2011-002 which can be found at
- http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3
Wed Feb 16 13:00:00 2011 - 1.8.3-0.7.rc3
-
- The Asterisk Development Team has announced the third release candidate of
- Asterisk 1.8.3. This release candidate is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.3-rc3 resolves the following issues in addition to
- those included in 1.8.3-rc1 and 1.8.3-rc2:
-
-
* Fix regression that changed behavior of queues when ringing a queue member.
- (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
-
-
* Resolve deadlock involving REFER.
- (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3
Fri Feb 11 13:00:00 2011 Jeffrey C. Ollie - 1.8.3-0.6.rc2
- Bump release to build for F15
Wed Feb 9 13:00:00 2011 Jeffrey C. Ollie - 1.8.3-0.5.rc2
- Remove isa macros
Wed Feb 9 13:00:00 2011 Jeffrey C. Ollie - 1.8.3-0.4.rc2
- Make library dependencies architecture specific
Mon Feb 7 13:00:00 2011 Fedora Release Engineering - 1.8.3-0.3.rc2
- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild
Wed Jan 26 13:00:00 2011 Jeffrey C. Ollie - 1.8.3-0.2.rc2
The Asterisk Development Team has announced the second release candidate of
Asterisk 1.8.3. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to
those included in 1.8.3-rc1:
* Resolve issue where no Music On Hold may be triggered when using
res_timing_dahdi.
(Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
by francesco_r, rfrantik, one47)
* Resolve a memory leak when the Asterisk Manager Interface is disabled.
(Reported internally by kmorgan. Patched by russellb)
* Reimplemented fax session reservation to reverse the ABI breakage introduced
in r297486.
(Reported internally. Patched by mnicholson)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2
Wed Jan 26 13:00:00 2011 Jeffrey C. Ollie - 1.8.3-0.1.rc1
-
- The Asterisk Development Team has announced the first release candidate of
- Asterisk 1.8.3. This release candidate is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.3-rc1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release candidate:
-
-
* Resolve duplicated data in the AstDB when using DIALGROUP()
- (Closes issue #18091. Reported by bunny. Patched by tilghman)
-
-
* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
- (Closes issue #18464. Reported, patched by IgorG)
-
-
* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
- unit tests for the function that does the parsing.
- (Closes issue #18350. Reported by gbour. Patched by Marquis)
-
-
* When using cdr_pgsql the billsec field was not populated correctly on
- unanswered calls.
- (Closes issue #18406. Reported by joscas. Patched by tilghman)
-
-
* Resolve memory leak in iCalendar and Exchange calendaring modules.
- (Closes issue #18521. Reported, patched by pitel. Tested by cervajs)
-
-
* This version of Asterisk includes the new Compiler Flags option
- BETTER_BACKTRACES which uses libbfd to search for better symbol information
- within both the Asterisk binary, as well as loaded modules, to assist when
- using inline backtraces to track down problems.
- (Patched by tilghman)
Wed Jan 26 13:00:00 2011 Jeffrey C. Ollie - 1.8.2.3-1
-
- The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.2.3 resolves the following issue:
-
-
* Reimplemented fax session reservation to reverse the ABI breakage introduced
- in r297486.
- (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by
- mnicholson)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.2.2-2
- Build with SRTP support
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.2.2-1
-
- The Asterisk Development Team has announced a release for the security issue
- described in AST-2011-001.
-
- Due to a failed merge, Asterisk 1.8.2.1 which should have included the security
- fix did not. Asterisk 1.8.2.2 contains the the changes which should have been
- included in Asterisk 1.8.2.1.
-
- This releases is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
- 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while
- in pedantic mode, which can cause a stack buffer to be made to overflow if
- supplied with carefully crafted caller ID information. The issue and resolution
- are described in the AST-2011-001 security advisory.
-
- For more information about the details of this vulnerability, please read the
- security advisory AST-2011-001, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2
-
- Security advisory AST-2011-001 is available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.2.1-1
-
- The Asterisk Development Team has announced security releases for the following
- versions of Asterisk:
-
-
* 1.4.38.1
-
* 1.4.39.1
-
* 1.6.1.21
-
* 1.6.2.15.1
-
* 1.6.2.16.1
-
* 1.8.1.2
-
* 1.8.2.1
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
- 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while
- in pedantic mode, which can cause a stack buffer to be made to overflow if
- supplied with carefully crafted caller ID information. The issue and resolution
- are described in the AST-2011-001 security advisory.
-
- For more information about the details of this vulnerability, please read the
- security advisory AST-2011-001, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.38.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1
-
- Security advisory AST-2011-001 is available at:
-
- http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.2-1
-
- The Asterisk Development Team has announced the release of Asterisk 1.8.2. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.2 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* \'sip notify clear-mwi\' needs terminating CRLF.
- (Closes issue #18275. Reported, patched by klaus3000)
-
-
* Patch for deadlock from ordering issue between channel/queue locks in
- app_queue (set_queue_variables).
- (Closes issue #18031. Reported by rain. Patched by bbryant)
-
-
* Fix cache of device state changes for multiple servers.
- (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
- by russellb)
-
-
* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
- instead of redirecting the call.
- (Closes issue #18171. Reported by: SantaFox)
- (Closes issue #18185. Reported by: kwemheuer)
- (Closes issue #18211. Reported by: zahir_koradia)
- (Closes issue #18230. Reported by: vmarrone)
- (Closes issue #18299. Reported by: mbrevda)
- (Closes issue #18322. Reported by: nerbos)
-
-
* Fix reloading of peer when a user is requested. Prevent peer reloading from
- causing multiple MWI subscriptions to be created when using realtime.
- (Closes issue #18342. Reported, patched by nivek.)
-
-
* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
- so res_jabber doesn\'t think there is already an XMPP connection sending
- device state. Also clean up CLI commands a bit.
- (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
-
-
* Don\'t crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
- setting peer->cdr = NULL, set it to not post.
- (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
-
-
* Fixes issue with outbound google voice calls not working. Thanks to az1234
- and nevermind_quack for their input in helping debug the issue.
- (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.1.1-1
-
- The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.1.1 resolves two issues reported by the community
- since the release of Asterisk 1.8.1.
-
-
* Don\'t crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
- setting peer->cdr = NULL, set it to not post.
- (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
-
-
* Fixes issue with outbound google voice calls not working. Thanks to az1234
- and nevermind_quack for their input in helping debug the issue.
- (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
-
- For a full list of changes in this release candidate, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1
Mon Jan 24 13:00:00 2011 Jeffrey C. Ollie - 1.8.1-1
-
- The Asterisk Development Team has announced the release of Asterisk 1.8.1. This
- release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- The release of Asterisk 1.8.1 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is a sample of the issues resolved in this release:
-
-
* Fix issue when using directmedia. Asterisk needs to limit the codecs offered
- to just the ones that both sides recognize, otherwise they may end up sending
- audio that the other side doesn\'t understand.
- (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
-
-
* Resolve issue where Party A in an analog 3-way call would continue to hear
- ringback after party C answers.
- (Patched by rmudgett)
-
-
* Fix playback failure when using IAX with the timerfd module.
- (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
-
-
* Fix problem with qualify option packets for realtime peers never stopping.
- The option packets not only never stopped, but if a realtime peer was not in
- the peer list multiple options dialogs could accumulate over time.
- (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
- jpeeler)
-
-
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
- invalid data.
- (Closes issue #18161. Reported by wdoekes. Patched by tilghman)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
Tue Jan 18 13:00:00 2011 Dennis Gilmore - 1.8.0-6
- dont package up the ices bits on el the client doesnt exist for us
Tue Jan 18 13:00:00 2011 Dennis Gilmore - 1.8.0-5
- dont build the 389 directory server package its not available on rhel6
Fri Dec 10 13:00:00 2010 Dennis Gilmore - 1.8.0-4
- dont always build AIS modules we dont have the BuildRequires on epel
Fri Oct 29 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-3
- Rebuild for new net-snmp.
Tue Oct 26 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-2
- Always build AIS modules
Thu Oct 21 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-1
- The Asterisk Development Team is proud to announce the release of Asterisk
- 1.8.0. This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/
-
- Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
- Term Support (LTS) release, similar to Asterisk 1.4. For more information about
- support time lines for Asterisk releases, see the Asterisk versions page.
-
- http://www.asterisk.org/asterisk-versions
-
- The release of Asterisk 1.8.0 would not have been possible without the support
- and contributions of the community. Since Asterisk 1.6.2, we\'ve had over 500
- reporters, more than 300 testers and greater than 200 developers contributed to
- this release.
-
- You can find a summary of the work involved with the 1.8.0 release in the
- sumary:
-
- http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
-
- A short list of available features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP channel driver
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
-
- For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
-
- Thank you for your continued support of Asterisk!
Mon Oct 18 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.8.rc5:
-
- The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform
- compatibility IPv6 changes. In addition, the availability of the English sound
- prompts with Australian accents has been added.
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
-
- For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5
-
- This release candidate contains fixes since the last release candidate as
- reported by the community. A sampling of the changes in this release candidate
- include:
-
-
* Additional fixups in chan_gtalk that allow outbound calls to both Google
- Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
- and stunaddr.
- (Closes issue #13971. Patched by dvossel)
-
-
* Resolve manager crash issue.
- (Closes issue #17994. Reported by vrban. Patchd by dvossel)
-
-
* Documentation updates for sample configuration files.
- (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)
-
-
* Resolve issue where faxdetect would only detect the first fax call in
- chan_dahdi.
- (Closes issue #18116. Reported by seandarcy. Patched by rmudgett)
-
-
* Resolve issue where a channel that is setup and torn down
*very
* quickly may
- not have the right call disposition or ${DIALSTATUS}.
- (Closes issue #16946. Reported by davidw. Review
- https://reviewboard.asterisk.org/r/740/)
-
-
* Set TCLASS field of IPv6 header when SIP QoS options are set.
- (Closes issue #18099. Reported by jamesnet. Patched by dvossel)
-
-
* Resolve issue where Asterisk could crash on shutdown when using SRTP.
- (Closes issue #18085. Reported by st. Patched by twilson)
-
-
* Fix issue where peers host port would be lost on a SIP reload.
- (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)
-
- A short list of available features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP channel driver
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
-
- For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
Fri Oct 8 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.7.rc3
- This release candidate contains fixes since the release candidate as reported by
- the community. A sampling of the changes in this release candidate include:
-
-
* Still build chan_sip even if res_crypto cannot be built (use, but not depend)
- (Reported by a user on the mailing list. Patched by tilghman)
-
-
* Get notifications for call files only when a file is closed, not when created
- (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
-
-
* Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
- expects the DTMF to arrive on the RTP stream and not via jingle DTMF
- signalling.
- (Patched by dvossel. Tested by malcolmd)
-
-
* Fixes to allow chan_gtalk to communicate with the Gmail web client.
- (Patched by phsultan and dvossel)
-
-
* Fix to GET DATA to allow audio to be streamed via an AGI.
- (Closes issue #18001. Reported by jamicque. Patched by tilghman)
-
-
* Resolve dnsmgr memory corruption in chan_iax2.
- (Closes issue #17902. Reported by afried. Patched by russell, dvossel)
-
- A short list of available features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP channel driver
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
-
- For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3
Wed Oct 6 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.6.rc2
- This release candidate contains fixes since the last beta release as reported by
- the community. A sampling of the changes in this release candidate include:
-
-
* Add slin16 support for format_wav (new wav16 file extension)
- (Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
-
-
* Fixes a bug in manager.c where the default configuration values weren\'t reset
- when the manager configuration was reloaded.
- (Closes issue #17917. Reported by lmadsen. Patched by bbryant)
-
-
* Various fixes for the calendar modules.
- (Patched by Jan Kalab.
- Reviewboard: https://reviewboard.asterisk.org/r/880/
- Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
- Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
-
-
* Add CHANNEL(checkhangup) to check whether a channel is in the process of
- being hung up.
- (Closes issue #17652. Reported, patched by kobaz)
-
-
* Fix a bug with MeetMe where after announcing the amount of time left in a
- conference, if music on hold was playing, it doesn\'t restart.
- (Closes issue #17408, Reported, patched by sysreq)
-
-
* Fix interoperability problems with session timer behavior in Asterisk.
- (Closes issue #17005. Reported by alexcarey. Patched by dvossel)
-
-
* Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
- determined to be one of the most significant bottlenecks in SIP registration
- processing. This patch improved the speed of an astdb load test by 50000%
- (yes, Fifty-Thousand Percent). On this particular load test setup, this
- doubled the number of SIP registrations the server could handle.
- (Review: https://reviewboard.asterisk.org/r/825/)
-
-
* Don\'t clear the username from a realtime database when a registration
- expires. Non-realtime chan_sip does not clear the username from memory when a
- registration expiries so realtime probably shouldn\'t either.
- (Closes issue #17551. Reported, patched by: ricardolandim. Patched by
- mnicholson)
-
-
* Don\'t hang up a call on an SRTP unprotect failure. Also make it more obvious
- when there is an issue en/decrypting.
- (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
- twilson)
-
-
* Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
-
- A short list of available features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP channel driver
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
-
- For a full list of changes in the current release candidate, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
Thu Sep 9 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.5.beta5
- This release contains fixes since the last beta release as reported by the
- community. A sampling of the changes in this release include:
-
-
* Fix issue where TOS is no longer set on RTP packets.
- (Closes issue #17890. Reported, patched by elguero)
-
-
* Change pedantic default value in chan_sip from \'no\' to \'yes\'
-
-
* Asterisk now dynamically builds the \"Supported\" header depending on what is
- enabled/disabled in sip.conf. Session timers used to always be advertised as
- being supported even when they were disabled in the configuration.
- (Related to issue #17005. Patched by dvossel)
-
-
* Convert MOH to use generic timers.
- (Closes issue #17726. Reported by lmadsen. Patched by tilghman)
-
-
* Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to
- Asterisk that changed the SSRC during bridges and masquerades broke SRTP
- functionality. Also broken was handling the situation where an incoming
- INVITE had more than one crypto offer.
- (Closes issue #17563. Reported by Alexcr. Patched by twilson)
-
- Asterisk 1.8 contains many new features over previous releases of Asterisk.
- A short list of included features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP Channel
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5
Tue Aug 24 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.4.beta4
- This release contains fixes since the last beta release as reported by the
- community. A sampling of the changes in this release include:
-
-
* Fix parsing of IPv6 address literals in outboundproxy
- (Closes issue #17757. Reported by oej. Patched by sperreault)
-
-
* Change the default value for alwaysauthreject in sip.conf to \"yes\".
- (Closes issue #17756. Reported by oej)
-
-
* Remove current STUN support from chan_sip.c. This change removes the current
- broken/useless STUN support from chan_sip.
- (Closes issue #17622. Reported by philipp2.
- Review: https://reviewboard.asterisk.org/r/855/)
-
-
* PRI CCSS may use a stale dial string for the recall dial string. If an
- outgoing call negotiates a different B channel than initially requested, the
- saved original dial string was not transferred to the new B channel. CCSS
- uses that dial string to generate the recall dial string.
- (Patched by rmudgett)
-
-
* Split _all_ arguments before parsing them. This fixes multicast RTP paging
- using linksys mode.
- (Patched by russellb)
-
-
* Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure
- data has valid CSV formatting. Also list the special CEL variables that are
- available for use in the mapping. There are also several other CEL fixes in
- this release.
- (Patched by russellb)
-
- Asterisk 1.8 contains many new features over previous releases of Asterisk.
- A short list of included features includes:
-
-
* Secure RTP
-
* IPv6 Support in the SIP Channel
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4
Wed Aug 11 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.3.beta3
-
- This release contains fixes since the last beta release as reported by the
- community. A sampling of the changes in this release include:
-
-
* Fix a regression where HTTP would always be enabled regardless of setting.
- (Closes issue #17708. Reported, patched by pabelanger)
-
-
* ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
- (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson)
-
-
* Support \"channels\" in addition to \"channel\" in chan_dahdi.conf.
- (https://reviewboard.asterisk.org/r/804)
-
-
* Fix parsing error in sip_sipredirect(). The code was written in a way that
- did a bad job of parsing the port out of a URI. Specifically, it would do
- badly when dealing with an IPv6 address.
- (Closes issue #17661. Reported by oej. Patched by mmichelson)
-
-
* Fix inband DTMF detection on outgoing ISDN calls.
- (Patched by russellb and rmudgett)
-
-
* Fixes issue with translator frame not getting freed. This issue prevented
- g729 licenses from being freed up.
- (Closes issue #17630. Reported by manvirr. Patched by dvossel)
-
-
* Fixed IPv6-related SIP parsing bugs and updated documention.
- (Reported by oej. Patched by sperreault)
-
-
* Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a
- list of a specified item. Matches up with FIELDQTY() and CUT().
- (Closes #17713. Reported, patched by gareth. Tested by tilghman)
-
- Asterisk 1.8 contains many new features over previous releases of Asterisk.
- A short list of included features includes:
-
-
* Secure RTP
-
* IPv6 Support
-
* Connected Party Identification Support
-
* Calendaring Integration
-
* A new call logging system, Channel Event Logging (CEL)
-
* Distributed Device State using Jabber/XMPP PubSub
-
* Call Completion Supplementary Services support
-
* Advice of Charge support
-
* Much, much more!
-
- A full list of new features can be found in the CHANGES file.
-
- http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
Mon Aug 2 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.2.beta2
- Rebuild against libpri 1.4.12
Mon Aug 2 14:00:00 2010 Jeffrey C. Ollie - 1.8.0-0.1.beta2
- Update to 1.8.0-beta2
- Disable building chan_misdn until compilation errors are figured out (https://issues.asterisk.org/view.php?id=14333)
- Start stripping tarballs again because Digium added MP3 code :(
Sat Jul 31 14:00:00 2010 Jeffrey C. Ollie - 1.6.2.10-1
-
- The following are a few of the issues resolved by community developers:
-
-
* Allow users to specify a port for DUNDI peers.
- (Closes issue #17056. Reported, patched by klaus3000)
-
-
* Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
- set.
- (Closes issue #16815. Reported, patched by rain)
-
-
* If there is realtime configuration, it does not get re-read on reload unless
- the config file also changes.
- (Closes issue #16982. Reported, patched by dmitri)
-
-
* Send AgentComplete manager event for attended transfers.
- (Closes issue #16819. Reported, patched by elbriga)
-
-
* Correct manager variable \'EventList\' case.
- (Closes issue #17520. Reported, patched by kobaz)
-
- In addition, changes to res_timing_pthread that should make it more stable have
- also been implemented.
-
- For a full list of changes in the current release, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
Wed Jul 14 14:00:00 2010 Jeffrey C. Ollie - 1.6.2.8-0.3.rc1
- Add patch to remove requirement on latex2html
Tue Jun 1 14:00:00 2010 Marcela Maslanova - 1.6.2.8-0.2.rc1
- Mass rebuild with perl-5.12.0
Tue May 4 14:00:00 2010 Jeffrey C. Ollie - 1.6.2.7-1
-
* Fix building CDR and CEL SQLite3 modules.
- (Closes issue #17017. Reported by alephlg. Patched by seanbright)
-
-
* Resolve crash in SLAtrunk when the specified trunk doesn\'t exist.
- (Reported in #asterisk-dev by philipp64. Patched by seanbright)
-
-
* Include an extra newline after \"Aliased CLI command\" to get back the prompt.
- (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright)
-
-
* Prevent segfault if bad magic number is encountered.
- (Issue #17037. Reported, patched by alecdavis)
-
-
* Update code to reflect that handle_speechset has 4 arguments.
- (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger,
- mmichelson)
-
-
* Resolve a deadlock in chan_local.
- (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2)
Mon May 3 14:00:00 2010 Jeffrey C. Ollie - 1.6.2.7-0.2.rc3
- Update to 1.6.2.7-rc3
Thu Apr 15 14:00:00 2010 Jeffrey C. Ollie - 1.6.2.7-0.1.rc2
- Update to 1.6.2.7-rc2
Fri Mar 12 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.6-1
- Update to final 1.6.2.6
-
- The following are a few of the issues resolved by community developers:
-
-
* Make sure to clear red alarm after polarity reversal.
- (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown,
- Chainsaw, mikeeccleston)
-
-
* Fix problem with duplicate TXREQ packets in chan_iax2
- (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel)
-
-
* Fix crash in app_voicemail related to message counting.
- (Closes issue #16921. Reported, tested by whardier. Patched by seanbright)
-
-
* Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts
- (Reported, Patched, and Tested by alecdavis)
-
-
* For T.38 reINVITEs treat a 606 the same as a 488.
- (Closes issue #16792. Reported, patched by vrban)
-
-
* Fix ConfBridge crash when no timing module is loaded.
- (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky)
-
- For a full list of changes in this releases, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6
Mon Mar 8 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.6-0.1.rc2
- Update to 1.6.2.6-rc2
Mon Mar 8 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.5-2
- Add a patch that fixes CLI history when linking against external libedit.
Thu Feb 25 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.5-1
- Update to 1.6.2.5
-
-
* AST-2010-002: Invalid parsing of ACL rules can compromise security
Thu Feb 18 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.4-1
- Update to 1.6.2.4
-
-
* AST-2010-002: This security release is intended to raise awareness
- of how it is possible to insert malicious strings into dialplans,
- and to advise developers to read the best practices documents so
- that they may easily avoid these dangers.
Wed Feb 3 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.2-1
- Update to 1.6.2.2
-
-
* AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
- remotely crash Asterisk by modifying the FaxMaxDatagram field of
- the SDP to contain either a negative or exceptionally large value.
- The same crash occurs when the FaxMaxDatagram field is omitted from
- the SDP as well.
Fri Jan 15 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.1-1
- Update to 1.6.2.1 final:
-
-
* CLI \'queue show\' formatting fix.
- (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by
- ppyy.)
-
-
* Fix misreverting from 177158.
- (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.)
-
-
* Fixes subscriptions being lost after \'module reload\'.
- (Closes issue #16093. Reported by jlaroff. Patched by dvossel.)
-
-
* app_queue segfaults if realtime field uniqueid is NULL
- (Closes issue #16385. Reported, Tested, Patched by haakon.)
-
-
* Fix to Monitor which previously assumed the file to write to did not contain
- pathing.
- (Closes issue #16377, #16376. Reported by bcnit. Patched by dant.
Tue Jan 12 13:00:00 2010 Jeffrey C. Ollie - 1.6.2.1-0.1.rc1
- Update to 1.6.2.1-rc1
Sat Dec 19 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-1
- Released version of 1.6.2.0
Wed Dec 9 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.16.rc8
- Update to 1.6.2.0-rc8
Wed Dec 2 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.15.rc7
- Update to 1.6.2.0-rc7
Tue Dec 1 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.14.rc6
- Change the logrotate and the init scripts so that Asterisk doesn\'t
try and write to / or /root
Thu Nov 19 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.13.rc6
- Make dependency on uw-imap conditional and some other changes to
make building on RHEL5 easier.
Fri Nov 13 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.12.rc6
- Update to 1.6.2.0-rc6
Mon Nov 9 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.11.rc5
- Update to 1.6.2.0-rc5
Fri Nov 6 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.10.rc4
- Update to 1.6.2.0-rc4
Tue Oct 27 13:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.9.rc3
- Add patch from upstream to fix how res_http_post forms paths.
Sat Oct 24 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.8.rc3
- Add an AST_EXTRA_ARGS option to the init script
- have the init script to cd to /var/spool/asterisk to prevent annoying message
Sat Oct 24 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.7.rc3
- Compile against gmime 2.2 instead of gmime 2.4 because the patch to convert the API calls from 2.2 to 2.4 caused crashes.
Fri Oct 9 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.6.rc3
- Require latex2html used in static-http documents
Wed Oct 7 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.5.rc3
- Change ownership and permissions on config files to protect them.
Tue Oct 6 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.4.rc3
- Update to 1.6.2.0-rc3
Wed Sep 30 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.3.rc2
- Merge firmware subpackage back into the main package.
Wed Sep 30 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.2.rc2
- Package internal help.
- Fix up some more paths in the configs so that everything ends up where we want them.
Wed Sep 30 14:00:00 2009 Jeffrey C. Ollie - 1.6.2.0-0.1.rc2
- Update to 1.6.2.0-rc2
- We no longer need to strip the tarball as it no longer includes non-free items.
Wed Sep 9 14:00:00 2009 Jeffrey C. Ollie - 1.6.1.6-2
- Enable building of API docs.
- Depend on version 1.2 or newer of speex
Sun Sep 6 14:00:00 2009 Jeffrey C. Ollie - 1.6.1.6-1
- Update to 1.6.1.6
- Drop patches that are too troublesome to maintain anymore or have been integrated upstream.
Tue Sep 1 14:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.26.rc1
- Add a patch from Quentin Armitage and rebuld.
Fri Aug 21 14:00:00 2009 Tomas Mraz - 1.6.1-0.25.rc1
- rebuilt with new openssl
Fri Jul 24 14:00:00 2009 Fedora Release Engineering - 1.6.1-0.24.rc1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild
Thu Mar 5 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.23.rc1
- Rebuild to pick up new AIS and ODBC deps.
- Update script that strips out bad content from tarball to do the
download and to check the GPG signature.
Mon Feb 23 13:00:00 2009 Fedora Release Engineering - 1.6.1-0.22.rc1
- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild
Sun Feb 8 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.21.rc1
- Update to 1.6.1-rc1
- Add backport of conference bridging that is slated for 1.6.2
- Add patches to conference bridging that implement CLI apps
Thu Jan 15 13:00:00 2009 Tomas Mraz - 1.6.1-0.13.beta4
- rebuild with new openssl
Sun Jan 4 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.12.beta4
- Fedora Directory Server compatibility patch/subpackage.
Sun Jan 4 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.10.beta4
- Fix up paths. BZ#477238
Sat Jan 3 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.9.beta4
- Update patches
Sat Jan 3 13:00:00 2009 Jeffrey C. Ollie - 1.6.1-0.8.beta4
- Update to 1.6.1-beta4
Tue Dec 9 13:00:00 2008 Jeffrey C. Ollie - 1.6.1-0.7.beta3
- Update to 1.6.1-beta3
Tue Dec 9 13:00:00 2008 Alex Lancaster - 1.6.1-0.6.beta2
- Rebuild for new gmime
Fri Nov 7 13:00:00 2008 Jeffrey C. Ollie - 1.6.1-0.5.beta2
- Add patch to fix missing variable on PPC.
Fri Nov 7 13:00:00 2008 Jeffrey C. Ollie - 1.6.1-0.4.beta2
- Update PPC systems don\'t have sys/io.h patch.
Fri Nov 7 13:00:00 2008 Jeffrey C. Ollie - 1.6.1-0.3.beta2
- PPC systems don\'t have sys/io.h
Fri Nov 7 13:00:00 2008 Jeffrey C. Ollie - 1.6.1-0.2.beta2
- Update to 1.6.1 beta 2
Wed Nov 5 13:00:00 2008 Jeffrey C. Ollie - 1.6.0.1-3
- Fix issue with init script giving wrong path to config file.
Thu Oct 16 14:00:00 2008 Jeffrey C. Ollie - 1.6.0.1-2
- Explicitly require dahdi-tools-libs to see if we can get this to build.
Fri Oct 10 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-1
- Update to final release.
Thu Sep 11 14:00:00 2008 - Bastien Nocera - 1.6.0-0.22.beta9
- Rebuild
Wed Jul 30 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.21.beta9
- Replace app_rxfax/app_txfax with app_fax taken from upstream SVN.
Tue Jul 29 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.20.beta9
- Bump release and rebuild with new libpri and zaptel.
Fri Jul 25 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.19.beta9
- Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011.
Fri Jul 25 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.18.beta9
- Add patch for LDAP extracted from upstream SVN (#442011)
Wed Jul 2 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.17.beta9
- Add patch that unbreaks cdr_tds with FreeTDS 0.82.
- Properly obsolete conference subpackage.
Thu Jun 12 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.16.beta9
- Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library.
Wed Jun 11 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.15.beta9
- Bump release and rebuild to fix libtds breakage.
Mon May 19 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.14.beta9
- Update to 1.6.0-beta9.
- Update patches so that they apply cleanly.
- Temporarily disable app_conference patch as it doesn\'t compile
- config/scripts/postgres_cdr.sql has been merged into realtime_pgsql.sql
- Re-add the asterisk-strip.sh script as a source file.
Tue Apr 22 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.13.beta8
- Update to 1.6.0-beta8
- Contains fixes for AST-2008-006 / CVE-2008-1897
Wed Apr 2 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.12.beta7.1
- Return to stripped tarballs since there\'s more non-free content in
the Asterisk tarballs than I thought.
Sun Mar 30 14:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.11.beta7.1
- Update to 1.6.0-beta7.1
- Update patches
- Back out some changes that were made because beta7 was tagged from
the wrong branch.
Fri Mar 28 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.10.beta7
- Update to 1.6.0-beta7
- The Asterisk tarball no longer contains the iLBC code, so we can
directly use the upstream tarball without having to modify it.
- Get rid of the asterisk-strip.sh script since it\'s no longer needed.
- Diable build of codec_ilbc and format_ilbc (these do not contain any
legally suspect code so they can be included in the tarball but it\'s
pointless building them).
- Update chan_mobile patch to fix API breakages.
- Add a patch to chan_usbradio to fix API breakages.
Thu Mar 27 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.9.beta6
- Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.
Tue Mar 25 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.8.beta6
- Update patches.
- Add patch to compile against external libedit rather than using the
in-tree version.
- Add -Werror-implicit-function-declaration to optflags.
- Get rid of hashtest and hashtest2 binaries that link to unfortified
versions of
*printf functions. They are compiled with -O0 which
somehow pulls in the wrong versions. These programs aren\'t
necessary to the operation of the package anyway.
Wed Mar 19 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.6.beta6
- Update to 1.6.0-beta6 to fix some security issues.
-
- AST-2008-002 details two buffer overflows that were discovered in
- RTP codec payload type handling.
-
* http://downloads.digium.com/pub/security/AST-2008-002.pdf
-
* All users of SIP in Asterisk 1.4 and 1.6 are affected.
-
- AST-2008-003 details a vulnerability which allows an attacker to
- bypass SIP authentication and to make a call into the context
- specified in the general section of sip.conf.
-
* http://downloads.digium.com/pub/security/AST-2008-003.pdf
-
* All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected.
-
- AST-2008-004 Logging messages displayed using the ast_verbose
- logging API call are not displayed as a character string, they are
- displayed as a format string.
-
* http://downloads.digium.com/pub/security/AST-2008-004.pdf
-
- AST-2008-005 details a problem in the way manager IDs are caculated.
-
* http://downloads.digium.com/pub/security/AST-2008-005.pdf
Tue Mar 18 13:00:00 2008 Tom \"spot\" Callaway - 1.6.0-0.5.beta5
- add Requires for versioned perl (libperl.so)
Wed Mar 5 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.4.beta5
- Update to 1.6.0-beta5
- Remove upstreamed patches.
Mon Mar 3 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.3.beta4
- Package the directory used to store monitor recordings.
Tue Feb 26 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.2.beta4
- Add patch from David Woodhouse that fixes building on PPC64.
Tue Feb 26 13:00:00 2008 Jeffrey C. Ollie - 1.6.0-0.1.beta4
- Update to 1.6.0 beta 4
Wed Feb 13 13:00:00 2008 Jeffrey C. Ollie - 1.4.18-1
- Update to 1.4.18.
- Use -march=i486 on i386 builds for atomic operations (GCC 4.3
compatibility).
- Use \"logger reload\" instead of \"logger rotate\" in logrotate file
(#432197).
- Don\'t explicitly specify a group in in the init script to prevent
Zaptel breakage (#426629).
- Split app_ices out to a separate package so that the ices package
can be required.
- pbx_kdeconsole has been dropped, don\'t specifically exclude it from
the build anymore.
- Update app_conference patch.
- Drop upstreamed libcap patch.
Wed Jan 2 13:00:00 2008 Jeffrey C. Ollie - 1.4.17-1
- Update to 1.4.17 to fix AST-2008-001.
Fri Dec 28 13:00:00 2007 Jeffrey C. Ollie - 1.4.16.2-1
- Update to 1.4.16.2
Sat Dec 22 13:00:00 2007 Jeffrey C. Ollie - 1.4.16.1-2
- Bump release and rebuild to fix broken dep on uw-imap.
Wed Dec 19 13:00:00 2007 Jeffrey C. Ollie - 1.4.16.1-1
- Update to the real 1.4.16.1.
Wed Dec 19 13:00:00 2007 Jeffrey C. Ollie - 1.4.16-2
- Add patch to bring source up to version 1.4.16.1 which will be
released shortly to fix some crasher bugs introduced by 1.4.16.
Tue Dec 18 13:00:00 2007 Jeffrey C. Ollie - 1.4.16-1
- Update to 1.4.16 to fix security bug.
Sat Dec 15 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-7
- Really, really fix the build problems on devel.
Sat Dec 15 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-6
- Tweaks to get to build on x86_64
Wed Dec 12 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-5
- Exclude PPC64
Wed Dec 12 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-4
- Don\'t build apidocs by default since there\'s a problem building on x86_64.
Tue Dec 11 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-3
- Really get rid of zero length map files.
Mon Dec 10 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-2
- Get rid of zero length map files.
- Shorten descriptions of voicemail subpackages
Fri Nov 30 13:00:00 2007 Jeffrey C. Ollie - 1.4.15-1
- Update to 1.4.15
Tue Nov 20 13:00:00 2007 Jeffrey C. Ollie - 1.4.14-2
- Fix license and other rpmlint warnings.
Mon Nov 19 13:00:00 2007 Jeffrey C. Ollie - 1.4.14-1
- Update to 1.4.14
Fri Nov 16 13:00:00 2007 Jeffrey C. Ollie - 1.4.13-7
- Add chan_mobile
Tue Nov 13 13:00:00 2007 Jeffrey C. Ollie - 1.4.13-6
- Don\'t build cdr_sqlite because sqlite2 has been orphaned.
- Rebase local patches to latest upstream SVN
- Update app_conference patch to latest from upstream SVN
- Apply post-1.4.13 patches from upstream SVN
Wed Oct 10 14:00:00 2007 Jeffrey C. Ollie - 1.4.13-1
- Update to 1.4.13
Tue Oct 9 14:00:00 2007 Jeffrey C. Ollie - 1.4.12.1-1
- Update to 1.4.12.1
Wed Aug 22 14:00:00 2007 Jeffrey C. Ollie - 1.4.11-1
- Update to 1.4.11
Fri Aug 10 14:00:00 2007 Jeffrey C. Ollie - 1.4.10.1-1
- Update to 1.4.10.1.
Tue Aug 7 14:00:00 2007 Jeffrey C. Ollie - 1.4.10-1 | |
